Sindicación del Foro Kunena http://www.elastix.org/ Sun, 19 May 2013 02:49:50 -0500 Kunena 1.5.11 http://www.elastix.org/components/com_kunena/template/default/images/english/emoticons/rss.gif http://www.elastix.org/ Sindicación del Foro Kunena http://www.elastix.org/index.php/en/component/kunena/6-translations/122095-die-kontrahentinnen-verkoerpern-unterschiedliche-frauenbilder-aufseiten-der-demokraten-tritt-michelle-obama-an.html#122095 Und auch wenn sich an diesem Tag bald alles um die Vorfreude auf die ebenfalls wieder kommende Nordseewoche dreht,http://www.blinkfang.de/hamburg.html (http://www.blinkfang.de/hamburg.html), um Sandburgenbauen, Muschelsuchen und Wellenschwimmen, das Experiment scheint für dieses Jahr geglückt. Dann liegen wieder die Uno-Karten auf dem Tisch. Da muss er jetzt durch,dass das einfach war. Sobald wir standen (http://www.xinphoto.com/ienglish/home/space.php?uid=5878&do=blog&id=170403),hollister online shop (http://www.billig-geht-immer.de/berlin.html), der Vater.. Die Kontrahentinnen verkörpern unterschiedliche Frauenbilder. Aufseiten der Demokraten tritt Michelle Obama an, Tochter eines schwarzen Maschinisten,Brüste knittern (http://travacle.com/user_blog.php), die in Harvard studierte und ihre Karriere trotz ihrer zwei Töchter weiterverfolgte. Ann Romney wiederum, Tochter eines weißen Unternehmers, Mutter von fünf Söhnen,ralph lauren outlet (http://www.blinkfang.de/schuhe.html), die noch nie in ihrem Leben selbst Geld verdiente,hollister online shop (http://www.billig-geht-immer.de/berlin.html), repräsentiert die konservativen Republikaner.. Der Traum spiegelt die Situation im Tod wie im Leben: Ihr Vater musste sein Haus verkaufen, das heißt einen Platz der Geborgenheit und Zugehörigkeit aufgeben,ralph lauren online shop (http://www.glashco-germany.de/bestellun.html), und auch nach seinem Ableben kann er nicht an dem dafür vorgesehenen Ort ruhen. Wenn Sie die wahren Umstände seines Todes nicht aufklären können, schlage ich Ihnen eventuell eine Familienaufstellung vor. Mit dieser sehr effektiven psychologischen Methode können Sie Licht ins Dunkel bringen. "Greenpeace hat die Wirkung des Verkaufs des Greenpeace-Magazins beim Discounter Lidl falsch eingeschätzt", sagt die Umweltschutzorganisation. Nach der aktuellen Ausgabe sollen keine Exemplare des Ökoblatts mehr bei dem Discounter erhältlich sein. Man beende den Verkauf des Magazins bei Lidl, "um jeglichen falschen Anschein zu vermeiden". Auf eine Zeitreise in die wilden 60er, die Swinging Sixties, entführt das diesjährige Geburtstagskind: Der Jaguar E-Type wird 50 Jahre jung. Die Besitzer der britischen Ikone freuen sich schon darauf, sich während des AvD-Oldtimer-Grand-Prix bei einem eigenen Rennen, bei der "Jaguar E-Type Challenge",auf dem Parkplatz und an der Stelle (http://www.wakeupforautism.com/cms/forum/topic/auf-dem-parkplatz-und-an-der-stelle?replies=1#post-673372), zu duellieren. Der Supersportwagen mit seiner unverwechselbaren und absolut kultigen Silhouette wurde zwischen 1961 und 1975 in rund 70.000 Exemplaren produziert,polo ralph aluren (http://www.glashco-germany.de/bestellun.html), er begeistert immer noch Menschen aller Generationen. Sun, 19 May 2013 02:24:04 -0500 http://www.elastix.org/index.php/en/component/kunena/116-security/122094-syrien-taiwan-tadsjikistan-zuerst-war-es-die-familie-priester.html#122094 Jeder kann wertvolle Zeit durch das Auffinden von Informationen sofort zu speichern. Speichern Reichtum durch Rückgewinnung wertvoller Organisation Space-Bereich sowie die Senkung Papierkosten. Halten Sie sich von der Handhabung umständlich Kopfschmerzen Papiere. Und da nach Mullin, wurde nun eindeutig festgestellt,oakley brillen (http://www.aquanet.de/opts.html), dass der Darm das Zentrum der Immunität ist, schlägt er regelmäßig isst Joghurt mit Probiotika,oakley frogskin (http://www.aquanet.de/opts.html), die Erhaltung einer gesunden Darmflora helfen. Sitzende Menschen sind eher als andere krank zu werden. Exerciseeven nur eine halbstündige zu einer Stunde walkinghas gezeigt worden, um Sie Funktion und zur Stärkung der Immunität.. Syrien. Taiwan. Tadsjikistan. Zuerst war es die Familie Priester, akribisch recherchieren würde die rishta oder Spiel für die Familie, indem sie auf seine immense Community Netzwerkressourcen. Dann war es eine Zeit der ehelichen Kleinanzeigen Spalten in allen führenden Zeitungen,ray ban aviator (http://www.maxondental.de/sportbrillen.html), die kurz und knapp über Kleinanzeigen zurückhaltend Bräute und Bräutigame schüchtern tragen würde. Heute werden Ehen buchstäblich in heavenonline gemacht, im Internet Raum.. Manche Frauen wollen Pflanzen und Gärten als ihre Hobbys wachsen. Männer auch gerne im Garten oder Hof als einen Weg, um weg von der Arbeit zu arbeiten. Es gibt etwas Entspannung über die Arbeit in den Schmutz oder Boden. Wenn Sie Schubladen hinzugefügt,oakley online shop (http://www.regina-halmich.org/templates.html), um sicherzustellen, Zubehör gehen direkt in den Schubladen nach Wäsche. Beim Wechsel der Jahreszeiten und ändern Sie beginnen, um Kleidung zu drehen, müssen Sie setzen außerhalb der Saison Kleidung in Ihrem Lagerplätze. Andernfalls führt nur dazu, zusätzliche Unordnung, die Sie festlegen können wieder auf die Spur zu einem unordentlichen Schrank.. Im Wesentlichen ist es ein Investmentfonds,oakley brillen (http://www.regina-halmich.org/templates.html), die an einer Börse gehandelt. So ETFs am Ende mit mehr Liquidität in den Stunden des Tages als eines Investmentfonds,ray ban shop (http://www.sundeka.de/sonnen.html), dass Sie in der Regel kann nur kommen, in die und aus der eine Zeit ein Tag. Sie handeln Börsengehandelte Fonds wie eine Aktie. Related articles: um die Klippen zu sehen ist der Gilde Park (http://eurovision-oem.com/eurovision-joins-mias-2012-amazing-experience.html#comment-28790) wenn sie don Sie lächeln (http://www.jiajinghb.com ews/html/?6636.html) können Sie verdienen sich eine psychiatrische Diagnose (http://www.0574zssm.com ews/html/?1859.html) Sun, 19 May 2013 01:43:54 -0500 http://www.elastix.org/index.php/en/component/kunena/27-miscellaneous/99757-add-sub-menu-item.html#122093 it's very useful and i send much thanks to you bro ... :) Sun, 19 May 2013 01:20:05 -0500 http://www.elastix.org/index.php/en/component/kunena/5-new-features/122092-integration-with-zimbra-8.html#122092 Hello all: Are there any plans to integrate Elastix with Zimbra 8? Regards, Dennis Ayala Zimbra New Features (http://www.zimbra.com/products/whats_new.html) Sun, 19 May 2013 00:49:53 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/122075-sccp-no-audio-for-calls-to-vm.html#122087 Hi. If you don't have no G729 codec, disable it. About iptables, sure, Elastix security module haven't any rules on port #2000. But in this case, no sound for everything. Sat, 18 May 2013 21:16:07 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/122078-h264-via-elastix.html#122086 bswash Can you post your version list from Elastix?? Can you also post your output from the Asterisk CLI - show peer Regards Bob Sat, 18 May 2013 19:02:25 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121845-sound-card-related-kernel-panic.html#122085 farzad741, I understand what you mean.... However, just in case someone has some ideas and allow further input....could you post the following from the linux commands lsusb (so we have the device number of the USB device) lspci (so we have the USB Controller in your machine) uname -a Also expand your own search on google and particularly look for the following usb centos kernel panic particularly you are looking for posts 1-2 years old.... Regards Bob Sat, 18 May 2013 18:55:38 -0500 http://www.elastix.org/index.php/en/component/kunena/38-hylafax/122041-incoming-fax-not-working-elastix-24.html#122084 velez, I have never had to use loopstart, but this may be something unique to your country. So we will assume it is working. The fact that you said that it is ringing twice..... Can you provide the part of /etc/asterisk/full log that has the call coming in and where it gets answered by the fax extension. Could you also provide the part of the hylafax log showing call trying to get through (if it is getting through to the IAX modem.... Regards Bob Sat, 18 May 2013 18:47:44 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121928-elastix-230-a-sperate-vtiger-540-server.html#122083 ukez, Lets go back to basics....lets confirm that your AMI can be seen 1) From your Elastix box Use telnet on port 5038 and connect to the local machine. You should see Asterisk Call Manager/1.1 in the terminal screen 2) From your Vtiger box use telnet on Port 5038 and connect to your Elastix box You should see Asterisk Call Manager/1.1 in the terminal screen If all the above are ok you have confirmed the following 1) AMI is running 2) AMI is not being blogged by firewalls 3) Networking is working correctly Next, just to make sure authentication is working, you want to login again try locally and then from your VTiger box I wont try and give you all the commands...but the following will give you the info you need to try authentication. http://www.scribd.com/doc/54892783/374/AMI-over-TCP Look for the section called AMI over TCP and If you get a little stuck, there are plenty of other articles on AMI over TCP (don't wander into AMI via HTTP - you'll end up on another tangent)....Also check things like manager.conf if you wish but don't edit them manually, this is all done via the GUI and works (otherwise half the rest of the tools would fail) If you still have issues post back here... Regards Bob Sat, 18 May 2013 18:37:33 -0500 http://www.elastix.org/index.php/en/component/kunena/68-endpoints/113791-experiencia-elastix-con-equipos-dinstar-fxofxs.html#122082 Amigo, saludo cordial, permitame hacerle una pregunta, yo llevé a cabo tu tutorial, las llamadas salen perfectamente , mi problema radica con las llamadas entrantes ya que cuando entran las recibe el ivr , pero cuando este las direcciona a la extencion de destino se cuelga la llamada o cuando se hace la marcacion directa de la extencion tambien se cuelga.. Sabes a que se debe??? version del gateway dinstar DWG2000 1G Sat, 18 May 2013 18:31:41 -0500 http://www.elastix.org/index.php/en/component/kunena/19-call-center/122057-call-center-gui.html#122081 Acevedo, I have not done it....but am aware of the GUI design. You could change the logo without any major issues, but unless you are well up on the GUI functionality which includes use of the php libraries that the Elastix GUI uses, you really would struggle (e.g. not just a weekend project). That said, and again I have not tried it, but you could look at what you could do with the ECCP http://www.elastix.org/index.php/en/component/content/article/67-newstest/558-eccp.html?Itemid=117 Worth a look Regards Bob Sat, 18 May 2013 18:19:12 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121712-elastix-database-.html#122080 makadarel, Just to confirm, are you using the Call Centre module?? or are you using the standard Elastix (as installed from the ISO?) Regards Bob Sat, 18 May 2013 18:14:37 -0500 http://www.elastix.org/index.php/en/component/kunena/27-miscellaneous/122079-bt--sip-trunk--voip-providers.html#122079 Hi, Anyone else using BT Business Infinity noticing SIP trunk registration problems ??? In the last couple of weeks I've been having trouble registering and or placing calls via my SIP trunk. At work and other locations it all works fine, just not on my BT connection. Was trawling the net and found this: http://www.theregister.co.uk/2013/04/30/bt_trolling_sip_in_battle_with_google/ http://www.btplc.com/Innovation/Licensing/Patentlicensingprogrammes/SIPTrunking/Formlicenceagreement.htm Is BT clamping down on SIP traffic / VOIP providers ??? Am considering moving to Virgin (sic) or another provider - any suggestions ??? Thanks, J :-) Sat, 18 May 2013 17:58:27 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/119917-chanspy-listening-in-to-all-but-these-extensions.html#122077 Itsm wrote: I have never seen a Chanspy configuration that actually works in order to filter extensions.. You can put a password, and even select through which exten. will it work from, but not to select to whom it may listen, I find FOP2 the best solution for those kind of things. Yeah I purchased FOP2 in the end and its money well spent if you ask me... I highly recommend it. Sat, 18 May 2013 17:15:45 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/89701-como-qresucitarq-telefono-cisco-7911-tras-reseteo.html#122067 Estimado, si ha encontrado alguna solución, favor de ayudarme que estamos en la misma situación...gracias Sat, 18 May 2013 08:50:46 -0500 http://www.elastix.org/index.php/en/component/kunena/86-question-installations/120726-impossible-de-se-connecter-a-linterface-asterisk.html?limit=10&start=20#122062 Ok merci de l'info et de l'avoir fait. Pour le suivi, je vais poster un mesage sur tout ticket de manière à être dans la boucle. Ne t'inquiètes pas pour çà. Bon week Sat, 18 May 2013 04:24:02 -0500 http://www.elastix.org/index.php/en/component/kunena/44-polycom/122055-polycom-vvx1500-support.html#122055 Has anyone integrated a Polycom VVX1500 with elastix 2.4? I have one on order and want to try it. Fri, 17 May 2013 22:54:54 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/118735-missing-0-from-isdn-caller-id.html#122049 dnlvry, It appears to be the editor you were using. There should be a hard return at the end of each line... Generally it is not a good idea to edit with the standard Windows editors as they will screw up formatting... either use nano (on the Linux console or SSH terminal program) or if you must edit on the Windows machine, then use WINSCP, where you can edit the files live on the Linux system, but they will pop up on the WINSCP editor which is like notepad or wordpad. Yes there is VI and many others but I have used these for many many years with few issues like the one you are having. I appreciate you are using windows editors to as you are familiar with them, but always good to edit with the ones above, less chance of mistakes....however having said that if you use NANO on long lines and don't understand what can happen when you edit long lines, it can cause you issues. WinSCP handles long lines well, and I know a lot of others who will not edit unless they use WINSCP, just to limit formatting mistakes. Regards Bob Fri, 17 May 2013 21:06:24 -0500 http://www.elastix.org/index.php/en/component/kunena/72-otros/121840-elastix-para-administrar-linea-analogica.html#122045 Muchas gracias por su respuesta. De hecho comprare una tarjeta análoga Digium, la intención es aplicar a esa línea todas las opciones posibles del servidor elastix. Saludos Fri, 17 May 2013 20:06:00 -0500 http://www.elastix.org/index.php/en/component/kunena/57-asterisk/122039-mi-asterisk-se-cuelga.html#122039 Les comento mi problema: troncalice aun servidor voip que me dio su ip y hostname mi cuenta sip mi usuario y contraseña, mi eslastix se logea el mometo que realizan una llamada hacia el numero de extension que troncalice realiza la llamada pero me dura la llamada 24 segundos y me aparece la siguinte advertencia en el CLI: [May 18 05:59:55] WARNING[584]: chan_sip.c:1967 retrans_pkt: Maximum retries exceeded on transmission 72d7456b470deff45482c4cc25f9c8ec@188.165.207.175:5060 for seqno 102 (Critical Response) [May 18 05:59:55] WARNING[584]: chan_sip.c:1989 retrans_pkt: Hanging up call 72d7456b470deff45482c4cc25f9c8ec@188.165.207.175:5060 - no reply to our critical packet. no se que puede ser porfavor ayuda gracias de antemano Fri, 17 May 2013 17:24:44 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/122038-conexion-entre-elastix-y-planta-alcatel-por-h323.html#122038 Hola. Estoy presentando un problema que no he logrado solucionar: Tengo un elastix en colombia y una planta alcatel en otro pais comunicados por un Lan to Lan. Hasta ahora, la comunicación entre la Alcatel y el Elastix va bien, pero entre el Elastix y la Alcatel presenta un problema durante el repique de la llamada, ya que se cae después de 3 segundos si la llamada no es contestada, el log me arroja el mensaje Everyone is busy/congested at this time (1:0/0/1) y escucho "todas las lineas están ocupadas", pero si contestan, la llamada se establece sin problemas. Espero alguien pueda colaborarme, a continuación les dejo mi configuración en el Elastix con su respectivo log ooh323.conf [general] port=1720 bindaddr=10.29.60.11 disallow=all allow=g729 dtmfmode=rfc2833 gatekeeper=DISABLE context=from-internal progress_setup=8 progress_alert=8 h245tunneling=yes [alcatel] type=friend context=from-internal host=10.29.0.14 port=1720 disallow=all allow=g729 canreinvite=yes dtmfmode=rfc2833 y el log.... == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Extension Changed 1140[ext-local] new state InUse for Notify User 1010 == Extension Changed 1140[ext-local] new state InUse for Notify User 1010 -- Executing [85069059@LocNacCelInt:1] Macro("SIP/1140-000005c0", "user-callerid,SKIPTTL,") in new stack -- Executing [s@macro-user-callerid:1] Set("SIP/1140-000005c0", "AMPUSER=1140") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1140-000005c0", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1140-000005c0", "1?Set(REALCALLERIDNUM=1140)") in new stack -- Executing [s@macro-user-callerid:4] Set("SIP/1140-000005c0", "AMPUSER=1140") in new stack -- Executing [s@macro-user-callerid:5] Set("SIP/1140-000005c0", "AMPUSERCIDNAME=Systems Engineer") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1140-000005c0", "0?report") in new stack -- Executing [s@macro-user-callerid:7] Set("SIP/1140-000005c0", "AMPUSERCID=1140") in new stack -- Executing [s@macro-user-callerid:8] Set("SIP/1140-000005c0", "CALLERID(all)="Systems Engineer" ") in new stack -- Executing [s@macro-user-callerid:9] ExecIf("SIP/1140-000005c0", "0?Set(CHANNEL(language)=)") in new stack -- Executing [s@macro-user-callerid:10] GotoIf("SIP/1140-000005c0", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] Set("SIP/1140-000005c0", "CALLERID(number)=1140") in new stack -- Executing [s@macro-user-callerid:20] Set("SIP/1140-000005c0", "CALLERID(name)=Systems Engineer") in new stack -- Executing [s@macro-user-callerid:21] NoOp("SIP/1140-000005c0", "Using CallerID "Systems Engineer" ") in new stack -- Executing [85069059@LocNacCelInt:2] NoOp("SIP/1140-000005c0", "Calling Out Route: SedeSanJose") in new stack -- Executing [85069059@LocNacCelInt:3] Set("SIP/1140-000005c0", "MOHCLASS=default") in new stack -- Executing [85069059@LocNacCelInt:4] Set("SIP/1140-000005c0", "_NODEST=") in new stack -- Executing [85069059@LocNacCelInt:5] Macro("SIP/1140-000005c0", "record-enable,1140,OUT,") in new stack -- Executing [s@macro-record-enable:1] GotoIf("SIP/1140-000005c0", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] ExecIf("SIP/1140-000005c0", "0?MacroExit()") in new stack -- Executing [s@macro-record-enable:5] GotoIf("SIP/1140-000005c0", "0?Group:OUT") in new stack -- Goto (macro-record-enable,s,15) -- Executing [s@macro-record-enable:15] GotoIf("SIP/1140-000005c0", "0?IN") in new stack -- Executing [s@macro-record-enable:16] ExecIf("SIP/1140-000005c0", "1?MacroExit()") in new stack -- Executing [85069059@LocNacCelInt:6] Macro("SIP/1140-000005c0", "dialout-trunk,4,9059,") in new stack -- Executing [s@macro-dialout-trunk:1] Set("SIP/1140-000005c0", "DIAL_TRUNK=4") in new stack -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1140-000005c0", "0?sub-pincheck,s,1") in new stack -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1140-000005c0", "0?disabletrunk,1") in new stack -- Executing [s@macro-dialout-trunk:4] Set("SIP/1140-000005c0", "DIAL_NUMBER=9059") in new stack -- Executing [s@macro-dialout-trunk:5] Set("SIP/1140-000005c0", "DIAL_TRUNK_OPTIONS=tTrW") in new stack -- Executing [s@macro-dialout-trunk:6] Set("SIP/1140-000005c0", "OUTBOUND_GROUP=OUT_4") in new stack -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1140-000005c0", "1?nomax") in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1140-000005c0", "0?skipoutcid") in new stack -- Executing [s@macro-dialout-trunk:10] Set("SIP/1140-000005c0", "DIAL_TRUNK_OPTIONS=tTW") in new stack -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1140-000005c0", "outbound-callerid,4") in new stack -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1140-000005c0", "0?Set(CALLERPRES()=)") in new stack -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1140-000005c0", "0?Set(REALCALLERIDNUM=1140)") in new stack -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1140-000005c0", "1?normcid") in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [s@macro-outbound-callerid:6] Set("SIP/1140-000005c0", "USEROUTCID=") in new stack -- Executing [s@macro-outbound-callerid:7] Set("SIP/1140-000005c0", "EMERGENCYCID=") in new stack -- Executing [s@macro-outbound-callerid:8] Set("SIP/1140-000005c0", "TRUNKOUTCID=") in new stack -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1140-000005c0", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1140-000005c0", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/1140-000005c0", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/1140-000005c0", "0?Set(CALLERID(all)=)") in new stack -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/1140-000005c0", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/1140-000005c0", "0?sub-flp-4,s,1") in new stack -- Executing [s@macro-dialout-trunk:13] Set("SIP/1140-000005c0", "OUTNUM=9059") in new stack -- Executing [s@macro-dialout-trunk:14] Set("SIP/1140-000005c0", "custom=AMP") in new stack -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1140-000005c0", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)tTW)") in new stack -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1140-000005c0", "dialout-trunk-predial-hook,") in new stack -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1140-000005c0", "") in new stack -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1140-000005c0", "0?bypass,1") in new stack -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1140-000005c0", "1?customtrunk") in new stack -- Goto (macro-dialout-trunk,s,22) -- Executing [s@macro-dialout-trunk:22] Set("SIP/1140-000005c0", "pre_num=AMP:OOH323/alcatel/") in new stack -- Executing [s@macro-dialout-trunk:23] Set("SIP/1140-000005c0", "the_num=OUTNUM") in new stack -- Executing [s@macro-dialout-trunk:24] Set("SIP/1140-000005c0", "post_num=") in new stack -- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/1140-000005c0", "1?outnum:skipoutnum") in new stack -- Goto (macro-dialout-trunk,s,26) -- Executing [s@macro-dialout-trunk:26] Set("SIP/1140-000005c0", "the_num=9059") in new stack -- Executing [s@macro-dialout-trunk:27] Dial("SIP/1140-000005c0", "OOH323/alcatel/9059,300,tTW") in new stack -- Called OOH323/alcatel/9059 -- OOH323/alcatel-61 is ringing == Everyone is busy/congested at this time (1:0/0/1) -- Executing [s@macro-dialout-trunk:28] NoOp("SIP/1140-000005c0", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 31") in new stack -- Executing [s@macro-dialout-trunk:29] Goto("SIP/1140-000005c0", "s-CHANUNAVAIL,1") in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1140-000005c0", "RC=31") in new stack -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1140-000005c0", "31,1") in new stack -- Goto (macro-dialout-trunk,31,1) -- Executing [31@macro-dialout-trunk:1] Goto("SIP/1140-000005c0", "continue,1") in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/1140-000005c0", "1?noreport") in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/1140-000005c0", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 31 - failing through to other trunks") in new stack -- Executing [continue@macro-dialout-trunk:4] Set("SIP/1140-000005c0", "CALLERID(number)=1140") in new stack -- Executing [85069059@LocNacCelInt:7] Macro("SIP/1140-000005c0", "outisbusy,") in new stack -- Executing [s@macro-outisbusy:1] Progress("SIP/1140-000005c0", "") in new stack -- Executing [s@macro-outisbusy:2] GotoIf("SIP/1140-000005c0", "0?emergency,1") in new stack -- Executing [s@macro-outisbusy:3] GotoIf("SIP/1140-000005c0", "0?intracompany,1") in new stack -- Executing [s@macro-outisbusy:4] GotoIf("SIP/1140-000005c0", "0?errado:next1") in new stack -- Goto (macro-outisbusy,s,9) -- Executing [s@macro-outisbusy:9] GotoIf("SIP/1140-000005c0", "0?busy:next2") in new stack -- Goto (macro-outisbusy,s,14) -- Executing [s@macro-outisbusy:14] Playback("SIP/1140-000005c0", "number-not-answering,noanswer") in new stack -- Playing 'number-not-answering.gsm' (language 'es') -- Executing [s@macro-outisbusy:15] Playback("SIP/1140-000005c0", "pls-try-call-later,noanswer") in new stack -- Playing 'pls-try-call-later.gsm' (language 'es') -- Executing [s@macro-outisbusy:16] NoOp("SIP/1140-000005c0", "HANGUPCAUSE=31") in new stack -- Executing [s@macro-outisbusy:17] Set("SIP/1140-000005c0", "CDR(userfield)= Hangupcause:31") in new stack -- Executing [s@macro-outisbusy:18] Hangup("SIP/1140-000005c0", "") in new stack == Spawn extension (macro-outisbusy, s, 18) exited non-zero on 'SIP/1140-000005c0' in macro 'outisbusy' == Spawn extension (LocNacCelInt, 85069059, 7) exited non-zero on 'SIP/1140-000005c0' -- Executing [h@LocNacCelInt:1] Macro("SIP/1140-000005c0", "hangupcall,") in new stack -- Executing [s@macro-hangupcall:1] GotoIf("SIP/1140-000005c0", "1?noautomon") in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] NoOp("SIP/1140-000005c0", "TOUCH_MONITOR_OUTPUT=") in new stack -- Executing [s@macro-hangupcall:4] GotoIf("SIP/1140-000005c0", "1?noautomon2") in new stack -- Goto (macro-hangupcall,s,6) -- Executing [s@macro-hangupcall:6] NoOp("SIP/1140-000005c0", "MONITOR_FILENAME=") in new stack -- Executing [s@macro-hangupcall:7] GotoIf("SIP/1140-000005c0", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,10) -- Executing [s@macro-hangupcall:10] GotoIf("SIP/1140-000005c0", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,13) -- Executing [s@macro-hangupcall:13] GotoIf("SIP/1140-000005c0", "1?theend") in new stack -- Goto (macro-hangupcall,s,15) -- Executing [s@macro-hangupcall:15] Hangup("SIP/1140-000005c0", "") in new stack == Spawn extension (macro-hangupcall, s, 15) exited non-zero on 'SIP/1140-000005c0' in macro 'hangupcall' == Spawn extension (LocNacCelInt, h, 1) exited non-zero on 'SIP/1140-000005c0' == Extension Changed 1140[ext-local] new state Idle for Notify User 1010 == Extension Changed 1140[ext-local] new state Idle for Notify User 1010 Fri, 17 May 2013 17:12:35 -0500 http://www.elastix.org/index.php/en/component/kunena/54-modulos/110997-modulo-monitoring-no-muestra-llamadas-de-queues.html?limit=10&start=10#122037 Yo tenia ese mismo problema corrientdo la 2.3 y segui la recomendacion en este foro de la actualizacion de elastix con el "[code]yum update elastix elstix-* asterisk asterisk-* dahdi dahdi-*[/code] pero al hacerlo, se afecto la parte de dahdi y ya no pude usar la T1 que tenia conectada. Termine por reinstalar toda la vesion 2.4 desde cero y con eso solucione el poder visualizar las llamadas de las colas en el monitoring y logico, los drivers de dahdi y su modulo se cargo fresquesito desde cero y ya me funciono todo. Lo malo es que tuve que reconfigurar todo de nuevo. Si en tu caso tu no tienes ninguna tarjeta que use dahdi, podras seguir trabajando sin problemas y tendras las grabaciones de las colas disponibles. Mi recomendacion es lo que yo hice; reinstalar Elastix 2.4 desde cero, ya que de esa manera obtienes el beneficio de tener las grabaciones y dahdi funcionando correctamente. Saludos. Fri, 17 May 2013 17:10:32 -0500 http://www.elastix.org/index.php/en/component/kunena/55-call-center/121964-callcenter.html#122035 No el problema no es de tiempo el problema es que cuando termina la llamada la opcion de agendar llamada no existe desaparece Fri, 17 May 2013 15:58:39 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/122018-procedencia-de-logs.html#122034 muy buen comando netsfk.. muchas gracias, y ahora.. como borro esos logs.. donde los encuentro... marcan 8.5 Gb me estan ocupando demasiado espacio en disco.. Fri, 17 May 2013 13:12:16 -0500 http://www.elastix.org/index.php/en/component/kunena/31-asterisk/121968-t1-not-working-anymore.html#122033 Hello. I ended up reinstalling Elastix 2.4 from scratch and it seems to have installed all the correct modules, as apparently it is working now. I still need to have an actual T1 connected to it but it is recognizing a loopback connector so far. Thank you for your response. Antonio. Fri, 17 May 2013 13:02:31 -0500 http://www.elastix.org/index.php/en/component/kunena/76-digium/122031-elastix-24-no-detecta-red-ni-tarjeta-fxs-fxo.html#122031 Saludos a todos, Soy nuevo con elastix y me de entrada me tropese con un problema al intalar la vestison elastix 2.4. No reconoce mi tarjeta de red ni la tarjeta Digium FXS FXO el comando lspci me dice que tarjeta de red es - ATHEROS Comunication Inc AR8132 fast ethernet - digium willcard TDM800 Alguien sabe donde podría conseguir los drivers adecuados?. gracias de antemano por cualquier sugerencia Fri, 17 May 2013 12:27:43 -0500 http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/120951-custom-trank-in-elastix-30.html#122028 If you can't do it, post it on: http://bugs.elasix.org Fri, 17 May 2013 12:16:31 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121993-conf-file-where-sip-trunk-is-stored.html#122026 If you create them through the web interface, they are created on sip_additionals.conf Fri, 17 May 2013 12:11:58 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/11930-hardware-optimo-para-server-elastix.html#122024 Puedes utilizar el mismo hardware que ya tienes, o si es que quieres usar un nuevo servidor, puedes enviar un correo a: sales@elastix.com solicitando un catálogo de nuestra línea de appliances con sus respectivas capacidades. Fri, 17 May 2013 12:09:42 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/114520-elastix-to-avaya-t1-trunk.html?limit=10&start=20#122022 Use ISDN trunk between them! Sangoma on your Elastix box -> DS1 card on your Avaya. This is how our Elastix production boxes works! H.323 is for Avaya only. Do not use the SIP service of Avaya as you will just waste of money for licensing. You know why? You will experience dead calls and robot calls. Lastly, Avaya is not good in SIP, it is not stable. Fri, 17 May 2013 12:05:03 -0500 http://www.elastix.org/index.php/en/component/kunena/84-le-coin-du-debutant/119458-affichage-du-numero-sda-de-lappellant.html?limit=10&start=10#122015 Fort possible. ok tiens nous au courant. Fri, 17 May 2013 08:52:37 -0500 http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/122013-move-monitoring-files-to-elastix-24.html#122013 I am upgrading from elastix 1.6-12 to elastix 2.4. I need to move my monitoring files over to the new machine. I have made a backup thru elastix 1.6 using the backup under the system tab. I then moved the file over to the new server and restored it. However, when I go to view the recordings on the new 2.4 server using the monitoring option under the pbx tab nothing is appearing. The filter is set to show 3 months and the files are located in the right directory /var/spool/asterisk/monitor and the owner and group are asterisk. All cdr files were imported over thru phpmayadmin. The monitoring files can be seen under recordings in freepbx just not elastix Fri, 17 May 2013 07:03:42 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121911-production-system-down-help-please.html#122010 READ THE MANY POSTS (Sorry for the capitals).... Before doing any upgrade to a production system perform an image backup!!!! Like any business, that is dependent on their IT Infrastructure, particularly the phones, it is common sense to either have a test system, or a standby system. Now I appreciate everyone has a budget, however there is no excuse for not doing an image backup of your system before upgrading major components. I am not perfect either. I made the mistake once, and never made it again....It is not worth the stress of rebuilding a system on the fly with a customer breathing down your back. Every system that is installed, before it goes live or as it goes live has an image backup done - stored on an External USB and from that day forward, weekly backups of the Freepbx config ftp'ed offsite. Never had a site that could not be restored, and generally in a time less than 30min....no stress... Regards Bob Fri, 17 May 2013 06:14:30 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121566-how-to-add-welcome-message-to-particular-extension.html#122006 Just to place some alternatives for those reading this thread. Step 1 - create a recording...(as you have already stated) Step 2 - create an announcement using that recording Step 3 - On extension e.g. 201 - "add a followme", on the followme, use a "initial ring time" of 0 (forcing it straight to the FollowMe for that extension). You now have the choice of putting the same extension in here e.g. 201 and a short ring time (if you still want it to ring the phone for a short period) or don't put an extension in. This will cause it to flow down straight to the "Destination if no Answer" Step 4 - Set "Destination if no Answer" to your announcement Why is this useful?? 1) It means that your incoming DID if pointing to this extension will go to the announcement 2) If someone rings internally it will go to the announcement Regards Bob Fri, 17 May 2013 05:11:14 -0500 http://www.elastix.org/index.php/en/component/kunena/79-general/121931-qtoutes-les-lignes-sont-occupeesq.html#122005 En analysant les logs Asterisk afin de vérifier ce qui c'est passé à l'instant T. Si le trunk c'est bien déconnecté....etc S c'est le cas, peut-être une petite coupures du WAN ou si c'est récurent (toutes les 30mn), possible que defaultexpiry soit à modifier (3600 à 1800). Si utilisation de DynDNS, c'est courant ce genre de problème. Il y a des choses à faire (regardes sur le forum). Après il n'y a que toi qui peut faire les investigations et analyser les choses. Fri, 17 May 2013 04:49:36 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/122004-redirect-calls-from-ivr-to-external-number.html#122004 hi I am new to Elastix, installed 2.4.0, basic setup is done by using Elastix Easy and Elastix without Tears. I configured IVR and redirected to extensions, when a call is from outside depending on the DTMF call is landing on configured Extension. My problem is when we press * that call must be transferred to another phone no. that no. is a physical phone. I had struck up with this problem from past one month and cant resolve this issue. I exactly don't know the technical words inside and the configuration files to. Could any one please guide me to resolve this. Thanks in Advance Regards Fri, 17 May 2013 04:45:09 -0500 http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/121922-opening-the-url-from-a-separate-workstation.html#122000 Thanks a lot, it worked. Fri, 17 May 2013 02:06:13 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121989-all-lines-are-busy---.html#121998 LINUX version 2.6.18 Elastix 2.3.0 R 5 Fri, 17 May 2013 01:53:40 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121753-relocation-ip-address-changed.html#121985 that is a direct public ip no router in between, that's why I am saying I can access to the server thru ssh means I can access to the ip address to the server without problems, but I cannot get the web UI working, anything I needed to do the make the webUI works for login Many Thanks Thu, 16 May 2013 22:19:34 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121983-yum-upgrade-or-yum-update-all-o-yum-update-elastix.html#121983 Hi All, I have a conceptual question regarding any Elastix installation (mainly the newest ones version 2.x) After a fresh install from the CD, which of the following CLI commands should we run to update/upgrade the system before any configs are put in: 1- Yum update -y 2- Yum update elastix 3- yum upgrade all 4- yum upgrade elastix I am trying to get to what is best practice and why? Thanks Thu, 16 May 2013 21:35:12 -0500 http://www.elastix.org/index.php/en/component/kunena/46-others/121871-cisco-spa-302d-cordlesswireless-dect-handset.html#121982 89 views and no replies??? Has anyone done this before? IF not, which is a good wireless solution? Thu, 16 May 2013 20:36:10 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/117589-solvedcant-get-queue-recordings.html#121980 jgutierrez: What part of the update actually takes care of the recordings part? Running the whole command: yum update elastix elastix-* asterisk asterisk-* dahdi dahdi-* will break dahdi and if you, like myself have a T1/E1 card in your box, it will not work anymore. I am reinstalling version 2.3 and I would like to know if I can just run "yum update elastix" and it would make the queue recordings appear in the monitor section. Thank you. Thu, 16 May 2013 19:56:08 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/121872-necesito-que-elastix-me-llame-a-mi-celular.html#121976 para que te llame en el cuadrito de folow poner tal como se marca en la troncal y agregando el simbolo de gato con eso llama al celular saludos ... pd: para eso necesitas una troncal obviamente con salida a celular saludos ... Thu, 16 May 2013 19:07:35 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/121874-una-linea-pstn-siempre-libre-desocupada.html#121975 No, la tenía en otra empresa una troncal con 10 lineas y el mismo numero. Lo que sucede es que le había dicho que si se puede hacer a la vicepresidente y ella me dijo que investigue. Con esto podré decirle que no es posible sin contratar este servicio con nuestro proveedor de lineas analogas. Muchas gracias. Thu, 16 May 2013 18:57:51 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121574-assistance-with-qall-circuits-are-busy-nowq.html#121974 Moderators, is this possible, I think it would be very useful to have it. Thank you. Thu, 16 May 2013 18:48:30 -0500 http://www.elastix.org/index.php/en/component/kunena/34-flash-operator-panel/121846-fop2-displays-only-extensions.html#121973 To keep youy guys in the loop, the reason for my FOP2 displaying only extensions is because it was the trial version and it only supports 15 buttons. As soon as I licensed it, it displayed everything... Nice... Thu, 16 May 2013 18:43:13 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121948-sip-trunk-addition-issues--help-.html#121970 donovanpl123, Welcome to Elastix. Click on "Version" at the top of your PBX GUI screen, wait for it to display a popup screen and give us the complete version & release numbers please. amphibian Thu, 16 May 2013 18:22:15 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/67752-aviso-de-disponibilidad-de-extension.html#121969 Buen dia: me tope con un problema similar, tendras aun el soft que ofreces ? Gracias un saludo Thu, 16 May 2013 17:04:22 -0500 http://www.elastix.org/index.php/en/component/kunena/54-modulos/121966-vtiger-en-elastix.html#121966 Hola, me gustaria saber si en la web hay algun documento que podamos usar para hacer una buena configuacion de vtiger con elastix. Thu, 16 May 2013 16:24:08 -0500 http://www.elastix.org/index.php/en/component/kunena/59-a2billing/117931-manual-de-a2billing-ver-194-espanol.html#121965 A2billing funciona correctamente pero le aconsejo actualizar a la ultima version. Thu, 16 May 2013 16:16:09 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/121939-incluir-en-grupo-de-timbrado-ext-de-otro-elastix.html#121963 Si no te funciona como te dijo jgutierrez prueba creando extensiones tipo other custom en el servidor A que apunten a las extensiones del servidor B Thu, 16 May 2013 15:08:26 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/120366-como-configurar-skype-en-elastix.html#121962 Saludos, Como puedo integrar skype conect en mi pbx elastix, me podrian ayudar con esto, se lo agradeceria de corazon Thu, 16 May 2013 13:59:01 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121957-divert-withheld-calls-to-recording.html#121957 Hello guys, can anyone tell me how to divert withheld inbound calls to a audio recording. Thu, 16 May 2013 13:34:16 -0500 http://www.elastix.org/index.php/en/component/kunena/5-new-features/5200-ldap-active-directory.html?limit=10&start=20#121955 Hi Good job and nice share. +1 Thu, 16 May 2013 12:28:33 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121942-need-to-upload-leads.html#121954 Wow. No one has an answer for one of the most common practices in all call center software. Thu, 16 May 2013 12:22:56 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121952-iax2-trunk-unable-to-create-channel-of-type-iax2.html#121952 Hi, Today we had an issue with our IAX2 trunk where it is now only one way communication. We have two Elastix 2.3 servers, one in the USA and the other in Europe. They are connected on the VPN and port 4569 open on each ends router. When we call from the USA to Europe via extension it works however from Europe to the USA we get all circuits are busy now. The log shows: Unable to create channel of type 'IAX2' (cause 20 - Subscriber absent) Any ideas. We haven't had any changes in either system, it just stopped working. Thanks in advance :) Thu, 16 May 2013 12:14:16 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/121827-usuarios-grupos-y-privilegios.html#121951 Intenta ingresar utilizado firefox o google chrome. Thu, 16 May 2013 11:47:56 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/121820-error-al-digitar-un-ext-de-otra-pbx.html#121950 Si es que la llamada ingresa a tu central a través de una troncal SIP, intenta ponerle en la troncal: dtmfmode=rfc2833 o dtmfmode=inband o dtmfmode=auto Thu, 16 May 2013 11:46:12 -0500 http://www.elastix.org/index.php/en/component/kunena/31-asterisk/121949-transfer-not-working-only-one-phone-shows-up.html#121949 Hello, I have a couple of issues. Elastix version is 2.4. We have Cisco 7960 phones. The two issues are as follows: ISSUE 1: When searching for phones under "PBX > Batch Configurations > Endpoint Configurator", only one phone shows up when I try to discover endpoints. It is not a *huge* deal as I can (and have been) just manually creating the MACADDRESS.cnf files by hand, but it would be nice to be able to configure all the phones via the endpoint configurator. ISSUE 2: The "transfer" button on the phones do not work. I am not sure if this is a hardware issue or a configuration issue. The phones do not seem to have any sort of web or telnet interface I can get in to What happens when someone attempts to transfer a call is the call is of course placed on hold, the user presses "transfer" but nothing happens. I have watched the asterisk log while this occurs ('asterisk -r -vvvvv') and the only thing that shows up in the log is the start of hold music when the transfer button is pressed, then the stop of hold music when the user came back to me to see if I had seen anything in the log. Any help or advice you might be able to offer ont he above would be appreciated. -SilkBC Thu, 16 May 2013 11:33:50 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121935-pbxconfig-blank-page.html#121935 Dear All, I am facing with pbxconfig blank page after trying to upgrade unembedded freepbx :( could anybody please help? .. trying the following: (but no luck) pear install DB [root@elastix ~]# pear install DB No releases available for package "pear.php.net/DB" Cannot initialize 'DB', invalid or missing package file Package "DB" is not valid install failed and [root@elastix bin]# ./retrieve_conf Added to globals: ASTETCDIR = /etc/asterisk Added to globals: ASTMODDIR = /usr/lib/asterisk/modules Added to globals: ASTVARLIBDIR = /var/lib/asterisk Added to globals: ASTAGIDIR = /var/lib/asterisk/agi-bin Added to globals: ASTSPOOLDIR = /var/spool/asterisk Added to globals: ASTRUNDIR = /var/run/asterisk Added to globals: ASTLOGDIR = /var/log/asterisk Added to globals: CWINUSEBUSY = true Added to globals: AMPMGRUSER = admin Added to globals: AMPMGRPASS = 3035lgi Added to globals: AMPDBENGINE = mysql Added to globals: AMPDBHOST = localhost Added to globals: AMPDBNAME = asterisk Added to globals: AMPDBUSER = asteriskuser Added to globals: AMPDBPASS = 3035lgi Added to globals: VMX_CONTEXT = from-internal Added to globals: VMX_PRI = 1 Added to globals: VMX_TIMEDEST_CONTEXT = Added to globals: VMX_TIMEDEST_EXT = dovm Added to globals: VMX_TIMEDEST_PRI = 1 Added to globals: VMX_LOOPDEST_CONTEXT = Added to globals: VMX_LOOPDEST_EXT = dovm Added to globals: VMX_LOOPDEST_PRI = 1 Added to globals: MIXMON_DIR = Added to globals: MIXMON_POST = Please update your modules and reload Asterisk by browsing to your server. [root@elastix bin]# Thu, 16 May 2013 08:14:35 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121932-outbound-calls-work-but-inbound-calls-fail.html#121932 Hi, I am new to Elastix and voip and try to get an inbound call to work. I have an Elastix PBX behind a firewall. I have a SIP trunk to a provider. I have both an outbound route and an inbound route. The inbound route goes directly to an extension. From a softphone, I can place a call to any destination. But when I call the DID to the same extension I just using for outbound, the incoming call gets an announcement from the provider as being unavailable. I see the call come to the softphone in the debug of the softphone but it doesn't ring. I have attached the sip debug. Any help would be much appreciated in interpreting the debug. Thanks. http://www.elastix.org/images/fbfiles/files/inbound_call_failure.txt Thu, 16 May 2013 07:36:13 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121930-audio-not-working-for-newly-created-elastix-pbx.html#121930 Dear Team, I am very new in elastix pbx system. My requirement is internal pbx system and no need to connect with phone lines. I have done following points for installation 1. I have assigned public and private ipaddress on newly installed box. 2. Created 5 general sip devices 3. Sussefully authenticated with 3CX and Xlite 4. Tried to call other extension and recieved incoming call 5. No audio for incoming and outgoing calls 6. Automatically disconnected after some seconds How I can sort out this problem Regards, Anzar Thu, 16 May 2013 07:17:38 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121929-iax2-trunk-between-two-elastix-servers.html#121929 Hi Guys. Im working in a small Company that Using elastix for all the phone and fax needs. i installed elastix on another machine outside the office(Abroad Elastix) for people going abroad and need phone services. i created an IAX2 trunk between the two machines which the main machine is User-details are context=from-trunk and the Abroad machine user-details are context - from-internal. i configure an extension in the abroad machine. the extension can dial outside using the main elastix, and can recieve calls from elastix main using a specific DID inbound route which i configured. the problem i having is that when Abroad extension try to dial outside to Main elastix the outside CID is the Default main elastix and not the CID which i defined in the Abroad extension. i hope you can help me solve this problem. thanks in advance. good day. Thu, 16 May 2013 07:16:29 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/120165-invalid-csv-file-in-the-elastix-24.html#121926 Hi, i think we are not talking about the same issue, but every line of the .CSV file should respect the form: Display Name,"User Extension","Direct DID","Outbound CID","Call Waiting","Secret","Voicemail Status","Voicemail Password","VM Email Address","VM Pager Email Address","VM Options","VM Email Attachment","VM Play CID","VM Play Envelope","VM Delete Vmail","Context","Tech" Exemple: unes,"101","","","DISABLED","101","enabled","123","","","","no","no","no","no","from-internal","sip" Good luck Thu, 16 May 2013 05:04:06 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/118905-sms-gsm-modem-elastix-and-android.html#121924 Please can you give me a guide on how you got Huawei E153 GSM modem running on Elastix 2.4 as a GSM gateway. Thank you. Thu, 16 May 2013 04:46:45 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/112721-como-configurar-rutas-salientes-y-entrantes.html#121923 Las rutas dependen de su plan de marcación (si marca 9 por la línea 1 por ej.) y luego del orden en caso de estar ocupados o no disponibles. Para diferenciar que una extensión salga por una ruta u otra marcando el mismo número, sólo es posible con el módulo de contextos de freepbx que yo sepa.También este aplica restricciones a cada extensión. también se puede crear un contexto pero es más complicado. Saludos Thu, 16 May 2013 03:05:37 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121826-implementationsupport-partner-list.html#121920 Parish_troy, Welcome to Elastix. What part of the US are you in? Me Texas. Being in the US also I started here looking for what you are requesting: http://www.elastix.org/index.php/en/reseller.html There is contact info so that should you need more info you can contact direct. Hope this helps, amphibian Thu, 16 May 2013 01:10:59 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121778-voicemail-configuration-and-voicemail-to-email.html#121919 raj2013, Welcome to Elastix. In order to obtain your desire request I have provided a step-by-step guide for you. Step-by-step guide: 1: Select "Product Info" tab at top of this web page, 2: then select "Manual/Books", 3: Then download "Elastix Without Tears", 4: Prepare to gain knowledge - Get big cup of coffee, some donuts, note pad, and a pen & paper, 5: Read "Elastix Without Tears" 6: Then configure voicemail and voicemail-to-email, 7: Finished, sit back and rejoice that you now have learned, setup, and can use Elastix with knowledge. amphibian Thu, 16 May 2013 01:01:00 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121817-issue-with-multiple-inbound-calls.html#121917 jrosetto, 1: What version of Elastix are you using? 2: Are you using a softphone, VoIp Desk phone or a Analog phone with a ATA? 3: Did you set up this machine with Elastix or someone else? If you set it up, then did you select call waiting on extension in question? amphibian Thu, 16 May 2013 00:44:39 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121834-virtual-sales.html#121916 kallee, If you have used asterisk before than I'm sure your are aware of "DISA" and "How to setup IVRs". What you want to do is doable from the Elastix GUI by setting up a IVR and DISA routes. Download and read the "Elastix Without Tears" located under "Product Info" tab on the site, then select "Manuals/Books", read, study, and be educated easily... amphibian Thu, 16 May 2013 00:38:22 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121858-add-one-more-phone-line.html#121914 skhan786, Tab at top of this webpage "Product Info", select "Manual/Books", download and print out or just read on your computer "Elastix without tears"... it will describ how to setup another landline inbound to your Elastix Server. amphibian Thu, 16 May 2013 00:30:31 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121873-basic-help-file-or-pdf.html#121913 macmann, Placing your mouse over an item normanlly pops a help/info box describing what the parameters rrequired are. For other helpful info download the "Elastix Without Tears" located under "Product Info", "Manual/Books"... Provides a lot of helpful info on Elastix. amphibian Thu, 16 May 2013 00:24:51 -0500 http://www.elastix.org/index.php/en/component/kunena/74-rhino/121910-avant-hier.html#121910 Le décryptage est un de ces vocables à la mode qui a envahi la langue journalistique. En principe, synonyme d'analyse ou d'explication, le terme est le plus souvent un cache-misère pour désigner n'importe quel commentaire. Et, comme nombre de leurs confrères, les journalistes du Lab baptisent eux-aussi leurs commentaires personnels du noble nom de décryptage Or ce qui les anime, c'est visiblement de ne rechercher que doubles discours, postures,adidas f50 pas cher (http://www.cognelec.fr/foots.html), ambitions, turpitudes et polémiques derrières les faits, les gestes et les déclarations des responsables politiques…. Le personnel vous offre un service d'assistance pour les visites touristiques/achat de billets au besoin. Une borne Internet est disponible. La réception dispose d'un coffre-fort et ses employés sont disponibles à des horaires limités,mercurial vapor (http://www.cognelec.fr/foots.html). Mediapart, récemment mis sous les feux de la rampe avec l Cahuzac,nike air max (http://www.necplusfrance.fr ecs.html), a aujourd publié un billet sur son blog pour informer ses abonnées d cyber-attaque. Le site explique que, depuis son récent succès, il a été victime de nombreuses tentatives de piratage qu a, jusqu lundi, réussi à repousser,air max 1 (http://www.necplusfrance.fr ecs.html). Cependant,nike free pas cher (http://www.chakma.org/free.html), avant-hier, une brèche a été percée dans la sécurité du média et il s retrouvé vulnérable de 12h53 à 15h49.. En 2009, Robert Charlebois met en sc le spectacle "Il une fois la bo chansons", un hommage aux grands auteurs-compositeurs-interpr qu tels Gilles Vigneault ou F Leclerc. Plus d'une centaine de repr sont donn au Qu l'artiste passe l' 2010 en studio, Montr pour l'enregistrement de son 21e album,nike free run (http://www.chakma.org/free.html), "Tout est bien". Sur ce disque sorti au Canada l'automne, Charlebois convoque violons, guitares, percussions, pianos pour orchestrer les douze chansons par lui-m Jean-Loup Dabadie et David McNeil ou adapter Mozart ou les lettres de Saint Augustin. Related articles: En Libye (http://ffdl.gotoip3.com ews/html/?2451.html) où Isabelle Carré fait du chocolat (http://kamaz-sochi.ru/workdisforum/topic/610339?replies=1#post-608055) l'album «Jambalaya» (http://bbs.lj8.com/showtopic-3777902.aspx) Wed, 15 May 2013 23:27:51 -0500 http://www.elastix.org/index.php/en/component/kunena/79-general/121830-elastix-et-la-base-de-donnees.html#121909 Il y a une truc qui m'épate..!! C'est que tu sois capable de coder une application et qu'il te soit très compliqué de comprendre une truc simple! http://www.elastix.org/images/fbfiles/images/http.png C'est ton appli qui donnera les information à Asterisk (Elastix) Toi tu ne déposes rien dans Asterisk! Tu dois faire une requête du style: http://ton_PC_avec_ton_appli/sous-dossier/index.php?num=0123456789 Et ton appli devra te retourner le Nom et Prénom correspondant au 0123456789 dans TA base de données locale sur TON PC. Rien de sorcier à comprendre! En plus tu as menu contextuel dans le CID LookUP Source qui te guide. Je ne vois pas où est le problème! Wed, 15 May 2013 23:00:14 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/119063-cambiar-idioma-a-espanol.html#121899 Que pena corrijo algo de lo anterior es: Language=es Wed, 15 May 2013 18:52:27 -0500 http://www.elastix.org/index.php/en/component/kunena/57-asterisk/82783-problemas-con-la-fecha-de-mail-de-voicemail.html#121897 Hola leonumark, Por favor me confirmas si lo pudiste solucionar y como lo lograstes. Gracias Wed, 15 May 2013 18:42:46 -0500 http://www.elastix.org/index.php/en/component/kunena/58-freepbx/25030-voicemail-attachment-a-mi-email.html#121896 Hola linuchero pues segun lo que veo es que no tiene un dominio conocido configurado en tu elastix, si quiere entra por ssh ----> setup ----> configuracion de la red ----> configuracion de dns ---> nombre del equipp ----> "Debes colocar algo comercial .com". Por otro tengo la mismo problem planteado por diegote, alguien me puede ayudar con ese mismo inconveniente. Saludos, JF Wed, 15 May 2013 17:48:59 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121854-where-is-qnever-override-calleridq-in-sip-trunk.html#121893 Hi jgutierrez, thank you for feedback and help. Already try this option - but my SIP provider still rejects forwarded calls - even if I use "Force Trunk CID". My root issue is is with "Call Forward No Answer/Unavailable". We have strange problem with "Call Forward No Answer/Unavailable" on Elastix. Elastix: 2.3.0 release 8 Asterisk: 1.8.20.0 release 0 Yealink phones T20P, T22P, T26P I can receive incoming (external) calls via SIP trunk without any problems. I can also place outgoing calls via SIP trunk without any problems. We have problem with redirecting (forward) calls to an external phone number (mobile number) if the called internal extension is not picking up (Call Forward No Answer / Unavailable). Example: Call comes from an external phone numbers via SIP trunk to internal extension (ext. 502 is destination number). If an internal ext. 502 is not answering within 10 seconds, we have set up call forwarding to a mobile number (via SIP trunk). As I can see in Elastix logs our proveider's SBC rejects the call. I do not know whether this is due to the security settings on the SBC or we had misconfigured the Elastix. Note: (below: Real phone numbers and IP are hidden due to security) == Using SIP RTP CoS mark 5 - Called SIP/OURSIP_Trunk/00xxyyyzzz - Got SIP response 484 "Address Incomplete" back from x.y.z.w:5060 == Everyone is busy / congested at this time (1:0 / 0/1) If we set unconditionally forward all calls (Call Forward All) on the phone (ext. 502)the phone, calls are easily diverted to a mobile number. Our SIP provider send us trace call from theirs SBC - but it's not much of a help. Help is appreciated. Wed, 15 May 2013 16:27:35 -0500 http://www.elastix.org/index.php/en/component/kunena/31-asterisk/121891-run-a-shell-command-on-asterisk-reload.html#121891 I'm trying to figure out a method to run a shell command when asterisk is reloaded. Either that or to run a shell command from the Elastix Web UI. Does anyone have any idea how to accomplish this. I have to believe this isn't completely out of the question. Thanks. Wed, 15 May 2013 16:05:38 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/121889-telefonos-h323.html#121889 Buenas! Como andan? Alguien me puede dar una pista de como configurar un telefono H323 en Elastix?? Desde ya, muchas racias Saludos Maxi Wed, 15 May 2013 15:21:20 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/121373-llamadas-pegadas.html#121888 Hola gracias, hare la pruebas. Saludos Wed, 15 May 2013 14:42:40 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/119552-conplejidad-de-la-password-extensiones.html?limit=10&start=10#121886 Saludos, Yo tambien aun sigo a la espera para ver si puedo resolver con esto de las password debiles, aun no se como modificar este archivo.. Wed, 15 May 2013 13:52:47 -0500 http://www.elastix.org/index.php/en/component/kunena/85-aide/121882-appel-call-center.html#121882 bonsoir a tous j'ai une seule question est-t'il possible à un utilisateur interne d'appeler un agent call center quand celui-ci est devant sa console d'agent en attente d'appel???? si oui dit moi comment configurer l'utilisateur interne pourqu'il puisse le joindre. merci pour vos réponses Wed, 15 May 2013 13:29:27 -0500 http://www.elastix.org/index.php/en/component/kunena/90-divers/121683-creation-ou-modification-template.html#121881 Pas de tuto. Sois curieux, regardes le contenu d'un fichier ./modules/un_module/index.php. Wed, 15 May 2013 13:26:59 -0500 http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/110172-uelastix.html?limit=10&start=20#121879 I have got it working by replacing few files on the boot partition Now I want to install G729a If you need my boot partition I can email to you Wed, 15 May 2013 12:46:32 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121877-queue-statistics-resets-on-reload.html#121877 Hello everyone :) I've made a very simple wallboard that shows the status of the queues on my system. It's based on a wallboard from here http://www.freepbx.org/trac/ticket/2280 The problem I'm running into now is that everytime I make some changes in the GUI of the Elastix and apply the settings(reload)the queue statistics reset. Is there a way around this? I've tried the keepstats = yes option in queues.conf with no luck. Best regards Björn Wed, 15 May 2013 11:58:22 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121870-restoring-backup-from-elastix-23x-on-elastix-24.html#121870 When i try to restore the backup it shows me the following message: "WARNING: Versions to restore are different. You could have any problem with the behavior of the programs after restoring. Press process to accept or cancel if not." Then if i try to proceed anyway it gives me the errors in the spoiler: Some of the extensions however get restored...about 5 from 80. Thank you all in advance for your help Wed, 15 May 2013 10:03:44 -0500 http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/121868-installation-issue-stops-during-install.html#121869 Nevermind, it works now. Wed, 15 May 2013 10:02:44 -0500 http://www.elastix.org/index.php/en/component/kunena/54-modulos/121804-modulo-roomx-precios-de-las-llamadas.html#121867 De nada. :) Wed, 15 May 2013 09:30:46 -0500 http://www.elastix.org/index.php/en/component/kunena/85-aide/121860-dial-rules-des-trunks-non-prises-en-compte.html#121865 Oui, pour effectivement différence entre 2.X et 1.X Wed, 15 May 2013 09:11:05 -0500 http://www.elastix.org/index.php/en/component/kunena/85-aide/121783-appels-indesirable-depuis-qunknown-peerq.html#121856 Ok, merci pour ces précisions intéressantes ! :) danardf wrote: 2 - Je n'ai pas nat=no dans les extensions du LAN. Ça pourrait améliorer la sécurité ? En quelque sorte oui. L'extension étant prévue pour être connectée en LAN, elle ne pourra donc par l'être depuis le WAN! Ok, donc c'est plus ou moins équivallent à : [code]deny=0.0.0.0/0.0.0.0 permit=192.168.1.0/255.255.255.0[/code] (en supposant, bien sûr, qu'on ait son LAN à cette adresse). Wed, 15 May 2013 06:58:45 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/55800-marcar-desde-ivr-extension-de-otro-elastix.html#121853 Muchas Gracias, Si funciona :D... Mi Elastix esta conectado via SIP a un Gateway Patton... Y PERFECTO Muchas Gracias... ---------------------------- mmayo: Crea un IVR y solo habilita el "Enable Direct Dial" y realiza los pasos descritos por jgutierrez (buscar el IVR deseado y la ruta destino)... Wed, 15 May 2013 04:19:19 -0500 http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/121849-elastix-30-alpha-web-logon-password.html#121849 I have done a new install of Elastix v3.0 Alpha and I have been able to logon to the linux interface using root but i cant logon to the Elastix web GUI I have tried all the usual logon usernames and passwords so what is the default username and password for Elastix 3.0 alpha. Wed, 15 May 2013 02:05:48 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121774-error-de-conexion-a-la-base-de-datos.html#121836 Who can help me…… Tue, 14 May 2013 20:37:30 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121524-does-elastix-hv-soln-to-allow-mobile-to-call-pstn.html#121835 Thanks Amphibian Tue, 14 May 2013 20:35:25 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/121756-problemas-aplicacion-goto-en-un-ivr.html#121832 Pega la salida del CLI (asterisk -r) para ver qué es lo que sucede Tue, 14 May 2013 18:14:27 -0500 http://www.elastix.org/index.php/en/component/kunena/19-call-center/75740-callcenter-module-outgoing-calls-not-working.html?limit=10&start=30#121831 i have follow all the instructions but when the agent log in for the outgoing campaing all it does is listen to music on hold i dont call here is the log tail -f /opt/elastix/dialer/dialerd.log [0] => Agent/1009 ) 2013-05-14 18:22:04 : (AMIEventProcess) DEBUG: AMIEventProcess::_pingAgente: Agent/1009 2013-05-14 18:22:04 : (ECCPProcess) DEBUG: ECCPConn::procesarPaquete: requerimiento pingagent procesado luego de (sec): 0.0016369819641113 Tue, 14 May 2013 17:25:59 -0500 http://www.elastix.org/index.php/en/component/kunena/33-a2billing/111595-freepbx-callback-and-a2billing.html?limit=10&start=50#121829 You're instruction works :-)) Thanks !!! Tue, 14 May 2013 17:09:22 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/121777-como-conectar-2-servidores-en-distintos-sitios.html#121823 Driveb escribió: hola muy buenas disculpen las molestias no se si ustedes me puedan ayudar con este error por favor me pueden ayudar tengo un problema al momento de marcar un numero y este tiene extensiones la centra no me permite digitar no me reconoce tengo instalado Elastix Tue, 14 May 2013 15:31:18 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121604-vmemailinc-keeps-getting-overwritten.html#121818 I made the file read only this morning and so far it has stayed and was not over-written. So far this has worked. I'll keep an eye on it. Thanks B) Tue, 14 May 2013 14:42:04 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/47786-cdr-reports-for-users-and-group-permissions.html?limit=10&start=10#121816 Thank you for your swift reply Claudio. I have already read those posts but Stelios solution looks to be a good, easy to apply option. I just couldn't see what line to add it on. I was hoping for the code that surrounds it to make it work. Thanks in advance Tue, 14 May 2013 14:12:37 -0500 http://www.elastix.org/index.php/en/component/kunena/19-call-center/121808-agent-console-login-error.html#121808 I followed the instructions online When I try to login to the console I get this error: Cannot start agent login - (internal) getagentstatus: Failed to authenticate to ECCP: Invalid username or password Have checked all my steps. I am, Stuck with no where to go. Suggestions? Tue, 14 May 2013 11:23:39 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121805-cid-name-prefix-and-queues.html#121805 Hello, I currently have an IVR set up where a user selects an option and is directed into a queue with a specific CID name prefix. I was wondering if it was possible to create a new IVR option which would direct the call into the same queue but change the CID name prefix displayed. I could create a different queue but it would require all my dynamic agents to sign in to two separate queues even though the membership will be exactly the same. Do I need to have two queues which users sign into individually or is there a workaround? I've tried creating another queue with a different prefix and making the original queue a member of it, but it only displays the CID name prefix of the original queue. Tue, 14 May 2013 11:02:05 -0500 http://www.elastix.org/index.php/en/component/kunena/31-asterisk/121803-outbound-circuits-are-full.html#121803 Hello I'm a bit new to asterik, it can receive calls using chan_mobile(Inbound), but when I try to do calls (Outbound) says circuits are full. And calls getting disconnect after that. log: extensions_override_elastix extensions_override_elastix sip_additional I kindly request help with this. Thanks you. ps: sorry for my english xD Tue, 14 May 2013 09:56:30 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/119678-problema-maquina-virtual-elastrix.html?limit=10&start=10#121802 ya solucione el problema de conexión, gracias si era así. adaptador puente qualcomm atheros AR Avanzadas PCnet-PCI II Permitir todo para entrar al servidor hay que desactivar el firewall entrar por la dirección que el nos da al principio elastix. "no por la que se puso en la instalación, después se cambia". user: admin pass: el que se puso en la instalación. pongo unas imágenes por si alguien las necesita. Tue, 14 May 2013 09:45:33 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121799-contacts-in-external-phonebook-problem.html#121799 Hi When we create contacts with the admin account in the external address phonebook they are only editable by the admin and not by other users who have permission to see edit the list, is there a way of making it so any/all users would be able to update this external phonebook. Tue, 14 May 2013 09:04:53 -0500 http://www.elastix.org/index.php/en/component/kunena/38-hylafax/121780-hylafax.html#121789 Hi Elastix has a wonderful implementation of Hylafax. https://IPELASTIX/index.php?menu=fax You also need to add iax2 device. Tue, 14 May 2013 04:38:07 -0500 http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/121788-problem-in-elastix.html#121788 hi all , please i need help in this issue i have one server elastix and i have also connected the softphones to this server directly , the problem is sometimes when i dial number it gives me strange tone . i attached this tone . http://www.4shared.com/music IcwSUF1/Sound_13.html Tue, 14 May 2013 04:37:36 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/84589-pbx-gui-is-not-working.html#121786 [root@ippbx ~]# df -h Sys. de fich. Tail. Occ. Disp. %Occ. Monté sur /dev/sdb1 70G 6,1G 61G 10% / /dev/sda1 99M 48M 47M 51% /boot tmpfs 688M 0 688M 0% /dev/shm [root@ippbx ~]# Tue, 14 May 2013 03:30:29 -0500 http://www.elastix.org/index.php/en/component/kunena/85-aide/121572-extensions-distantes--pb-de-port-et-pas-de-son.html#121784 Salut, Merci pour ton aide. Le problème semble bien venir du routeur : il n'est pas encore résolu, mais une mise à jour du firmware pour la dernière version dispo chez Netgear a bien changé les symptômes. On a encore des problèmes de son qui passent dans un sens et pas dans l'autre, mais toujours aussi ce problème de port 1024 qui, cette fois, est sur le softphone et plus sur le ST2030 !!! Nous avons cependant un autre problème pour lequel j'ouvre un nouveau fil (http://www.elastix.org/index.php/en/component/kunena/85-aide/121783-appels-indesirable-depuis-qunkno), puisque totalement différent (mais apparu lorsque j'ai modifié la config NAT...) Tue, 14 May 2013 03:22:32 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121280-showing-particular-phone-number-when-making-a-call.html#121776 Hi Thanks for your reply. Tue, 14 May 2013 00:48:14 -0500 http://www.elastix.org/index.php/en/component/kunena/31-asterisk/120311-unknown-inbound-caller-id.html#121770 Hello I have a slightly different situation, I can see the caller ID in the reports but on the telephone screen it apears "unknown" Analog lines in whose the caller ID works properly are connected to a Grandstream FXO gateway I hope for help. Mon, 13 May 2013 19:39:56 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121491-remote-site-and-other-questions.html#121767 +1 on Amphibians answer Mon, 13 May 2013 17:09:09 -0500 http://www.elastix.org/index.php/en/component/kunena/116-security/121765-users-disconnection.html#121765 Hi ALL I am a new comer with Elastix and I have a serious problem. I could not connect with any users created on my Elastix IPPBX even the admin. I succeeded to reinitlise the admin, but I keep on losing it. Have I been hacked? How can I solve this issue? Thanks for helping hand. Regards Mon, 13 May 2013 16:56:08 -0500 http://www.elastix.org/index.php/en/component/kunena/42-others/121164-atcom-ax-4s-4-isdn-card-detection-problem.html#121763 J made a fresh install of elastix 2.3 and my AX4B works fine. after a yum update its works also, but after push button detect hardware j d'ont see my card. test to morrow with 2.4. thanks pat Mon, 13 May 2013 14:10:30 -0500 http://www.elastix.org/index.php/en/component/kunena/116-security/121761-threat-or-bug.html#121761 Hello , dear security gurus. Today we started seeing this kind of warnings and code. Could a borrow a quick glance of your experienced minds... Is this a security issue or just some code failing? comments and pointers appreciated. thanks [code][May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-007-7506345' tries to include nonexistent context 'outrt-007-7506345-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-006-7506528' tries to include nonexistent context 'outrt-006-7506528-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-005-LDI' tries to include nonexistent context 'outrt-005-LDI-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-004-LDN' tries to include nonexistent context 'outrt-004-LDN-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-003-Celular044' tries to include nonexistent context 'outrt-003-Celular044-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-002-9_outside' tries to include nonexistent context 'outrt-002-9_outside-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-001-01900' tries to include nonexistent context 'outrt-001-01900-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outbound-allroutes' tries to include nonexistent context 'outbound-allroutes-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'from-did-direct-ivr' tries to include nonexistent context 'from-did-direct-ivr-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-local' tries to include nonexistent context 'ext-local-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-8' tries to include nonexistent context 'macro-from-zaptel-8-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-7' tries to include nonexistent context 'macro-from-zaptel-7-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-6' tries to include nonexistent context 'macro-from-zaptel-6-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-5' tries to include nonexistent context 'macro-from-zaptel-5-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-4' tries to include nonexistent context 'macro-from-zaptel-4-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-3' tries to include nonexistent context 'macro-from-zaptel-3-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-2' tries to include nonexistent context 'macro-from-zaptel-2-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-1' tries to include nonexistent context 'macro-from-zaptel-1-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-did-catchall' tries to include nonexistent context 'ext-did-catchall-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-did' tries to include nonexistent context 'ext-did-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-did-0002' tries to include nonexistent context 'ext-did-0002-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-did-0001' tries to include nonexistent context 'ext-did-0001-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-test' tries to include nonexistent context 'ext-test-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'app-chanspy' tries to include nonexistent context 'app-chanspy-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'app-zapbarge' tries to include nonexistent context 'app-zapbarge-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'app-pickup' tries to include nonexistent context 'app-pickup-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'app-userlogonoff' tries to include nonexistent context 'app-userlogonoff-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found [May 13 11:33:10] NOTICE[3642]: chan_iax2.c:9186 __iax2_poke_noanswer: Peer '3072' is now UNREACHABLE! Time: 2 [May 13 11:33:20] NOTICE[3650]: chan_iax2.c:8264 socket_process: Peer '3072' is now REACHABLE! Time: 3 -- Starting simple switch on 'DAHDI/4-1' -- Executing [s@from-pstn:1] Set("DAHDI/4-1", "__FROM_DID=s") in new stack -- Executing [s@from-pstn:2] Gosub("DAHDI/4-1", "app-blacklist-check|s|1") in new stack -- Executing [s@app-blacklist-check:1] LookupBlacklist("DAHDI/4-1", "") in new stack -- Executing [s@app-blacklist-check:2] GotoIf("DAHDI/4-1", "0?blacklisted") in new stack -- Executing [s@app-blacklist-check:3] Return("DAHDI/4-1", "") in new stack -- Executing [s@from-pstn:3] ExecIf("DAHDI/4-1", "1 |Set|CALLERID(name)=3336060867") in new stack -- Executing [s@from-pstn:4] SetMusicOnHold("DAHDI/4-1", "setramex2") in new stack -- Executing [s@from-pstn:5] Set("DAHDI/4-1", "__MOHCLASS=setramex2") in new stack -- Executing [s@from-pstn:6] Set("DAHDI/4-1", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [s@from-pstn:7] SetCallerPres("DAHDI/4-1", "allowed_not_screened") in new stack -- Executing [s@from-pstn:8] Goto("DAHDI/4-1", "timeconditions|1|1") in new stack -- Goto (timeconditions,1,1) -- Executing [1@timeconditions:1] GotoIfTime("DAHDI/4-1", "09:00-18:30|mon-fri|*|*?ivr-4|s|1") in new stack -- Goto (ivr-4,s,1) -- Executing [s@ivr-4:1] Set("DAHDI/4-1", "MSG=custom/setramex_conmutador_intro3") in new stack -- Executing [s@ivr-4:2] Set("DAHDI/4-1", "LOOPCOUNT=0") in new stack -- Executing [s@ivr-4:3] Set("DAHDI/4-1", "__DIR-CONTEXT=default") in new stack -- Executing [s@ivr-4:4] Set("DAHDI/4-1", "_IVR_CONTEXT_ivr-4=") in new stack -- Executing [s@ivr-4:5] Set("DAHDI/4-1", "_IVR_CONTEXT=ivr-4") in new stack -- Executing [s@ivr-4:6] GotoIf("DAHDI/4-1", "0?begin") in new stack -- Executing [s@ivr-4:7] Answer("DAHDI/4-1", "") in new stack -- Executing [s@ivr-4:8] Wait("DAHDI/4-1", "1") in new stack -- Executing [s@ivr-4:9] Set("DAHDI/4-1", "TIMEOUT(digit)=3") in new stack -- Digit timeout set to 3 -- Executing [s@ivr-4:10] Set("DAHDI/4-1", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing [s@ivr-4:11] Set("DAHDI/4-1", "__IVR_RETVM=") in new stack -- Executing [s@ivr-4:12] ExecIf("DAHDI/4-1", "1|Background|custom/setramex_conmutador_intro3") in new stack [May 13 11:33:24] WARNING[24804]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 -- Playing 'custom/setramex_conmutador_intro3' (language 'en') -- Executing [0@ivr-4:1] DBdel("DAHDI/4-1", "") in new stack -- Executing [0@ivr-4:2] Set("DAHDI/4-1", "__NODEST=") in new stack -- Executing [0@ivr-4:3] Goto("DAHDI/4-1", "ext-group|6007|1") in new stack -- Goto (ext-group,6007,1) -- Executing [6007@ext-group:1] Macro("DAHDI/4-1", "user-callerid|") in new stack -- Executing [s@macro-user-callerid:1] Set("DAHDI/4-1", "AMPUSER=3336060867") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("DAHDI/4-1", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("DAHDI/4-1", "1|Set|REALCALLERIDNUM=3336060867") in new stack -- Executing [s@macro-user-callerid:4] Set("DAHDI/4-1", "AMPUSER=") in new stack -- Executing [s@macro-user-callerid:5] Set("DAHDI/4-1", "AMPUSERCIDNAME=") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/4-1", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("DAHDI/4-1", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("DAHDI/4-1", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("DAHDI/4-1", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [s@macro-user-callerid:19] NoOp("DAHDI/4-1", "Using CallerID "3336060867" ") in new stack -- Executing [6007@ext-group:2] GotoIf("DAHDI/4-1", "1?skipdb") in new stack -- Goto (ext-group,6007,4) -- Executing [6007@ext-group:4] Set("DAHDI/4-1", "__NODEST=") in new stack -- Executing [6007@ext-group:5] Set("DAHDI/4-1", "__BLKVM_OVERRIDE=BLKVM/6007/DAHDI/4-1") in new stack -- Executing [6007@ext-group:6] Set("DAHDI/4-1", "__BLKVM_BASE=6007") in new stack -- Executing [6007@ext-group:7] Set("DAHDI/4-1", "DB(BLKVM/6007/DAHDI/4-1)=TRUE") in new stack -- Executing [6007@ext-group:8] Set("DAHDI/4-1", "RRNODEST=") in new stack -- Executing [6007@ext-group:9] Set("DAHDI/4-1", "__NODEST=6007") in new stack -- Executing [6007@ext-group:10] Set("DAHDI/4-1", "RecordMethod=Group") in new stack -- Executing [6007@ext-group:11] Macro("DAHDI/4-1", "record-enable|3006|Group") in new stack -- Executing [s@macro-record-enable:1] GotoIf("DAHDI/4-1", "1?check") in new stack -- Goto (macro-record-enable,s,4) -- Executing [s@macro-record-enable:4] AGI("DAHDI/4-1", "recordingcheck|20130513-113328|1368462801.2231") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck -- AGI Script recordingcheck completed, returning 0 -- Executing [s@macro-record-enable:5] MacroExit("DAHDI/4-1", "") in new stack -- Executing [6007@ext-group:12] Set("DAHDI/4-1", "RingGroupMethod=ringall") in new stack -- Executing [6007@ext-group:13] Macro("DAHDI/4-1", "dial|25|tr|3006") in new stack -- Executing [s@macro-dial:1] GotoIf("DAHDI/4-1", "0?dial") in new stack -- Executing [s@macro-dial:2] SetMusicOnHold("DAHDI/4-1", "setramex2") in new stack -- Executing [s@macro-dial:3] AGI("DAHDI/4-1", "dialparties.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi dialparties.agi: Starting New Dialparties.agi == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_additional.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found [May 13 11:33:28] WARNING[24807]: config.c:765 process_text_line: Unknown directive '#permit=192.168.0.0/255.255.255.0' at line 18 of /etc/asterisk/manager_custom.conf == Manager 'admin' logged on from 127.0.0.1 dialparties.agi: Caller ID name is '3336060867' number is '3336060867' dialparties.agi: Methodology of ring is 'ringall' -- dialparties.agi: Added extension 3006 to extension map -- dialparties.agi: Extension 3006 cf is disabled -- dialparties.agi: Extension 3006 do not disturb is disabled dialparties.agi: ExtensionState: 0 dialparties.agi: Extension 3006 has ExtensionState: 0 -- dialparties.agi: Checking CW and CFB status for extension 3006 -- dialparties.agi: dbset CALLTRACE/3006 to 3336060867 -- dialparties.agi: Filtered ARG3: 3006 == Manager 'admin' logged off from 127.0.0.1 -- AGI Script dialparties.agi completed, returning 0 -- Executing [s@macro-dial:7] Dial("DAHDI/4-1", "SIP/3006|25|trM(auto-blkvm)") in new stack [May 13 11:33:28] NOTICE[24804]: app_dial.c:1185 dial_exec_full: Hey! chan DA[/code] Mon, 13 May 2013 13:47:21 -0500 http://www.elastix.org/index.php/en/component/kunena/54-modulos/115159-base-de-datos-postgresql--ivr.html#121760 holaaaa JF como te ha ido? te comento que no he realizado la conexion de la BD postgres con el elastix no se programar en php, asi que opte por instalar postgres en el elastix y cree las tablas necesarias y hasta aqui todo esta bien perooooo desde el custon quiero hacer la conexion de esta forma include => IVR1 [globals] ;——– VARIABLES Telefono= total= [IVR1] exten => _69411,1,Playback(custom/cedula) exten => _69411,2,Read(Telefono,,8) ; variable telefono recupera lo q el usuario ingreso 8 caracteres exten => _69411,3,PGSQL(Connect conn localhost root contraseña Postgres) ; usuario: root, exten => _69411,4,PGSQL(Query resultid $ SELECT Nombre from CLIENTES where TELEFONO=$) ; consulta tabla CNT, campo idfono exten => _69411,5,PGSQL(Fetch fetchid $ suu) ; recupera el valor exten => _69411,6,Set(total=$) ; guarda en la variable total exten => _69411,7,PGSQL(Clear $) ;limpia exten => _69411,8,PGSQL(Disconnect $) ;desconecta exten => _69411,9,GotoIf($[$)} = 0]?11:20) ; si la consulta retorna valores vacios exten => _69411,10,Goto(11) exten => _69411,11,Playback(es/gracias) ; se le dice al usuario que no hubo datos de su consulta exten => _69411,12,Goto(23) exten => _69411,20,Playback(es/gracias) exten => _69411,21,SayNumber($) ; exten => _69411,22,Playback(es/gracias) exten => _69411,23,BackGround(custom/pregunta) ; si desea hacer otra consulta exten => _69411,24,WaitExten(3) exten => 1,1,Goto(IVR1,s,1) exten => _69411,1,Goto(IVR,s,4) exten => _69411,1,Hangup y no realiza la conexion, ps ando aun en eso :( :( Mon, 13 May 2013 13:42:21 -0500 http://www.elastix.org/index.php/en/component/kunena/57-asterisk/112396-al-recibir-fax-genera-fichero-core-y-da-error.html#121759 Hola Tuviste alguna solucion a tu problema? Ahora el nuestro esta mandando msgs similares. gracias [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-007-7506345' tries to include nonexistent context 'outrt-007-7506345-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-006-7506528' tries to include nonexistent context 'outrt-006-7506528-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-005-LDI' tries to include nonexistent context 'outrt-005-LDI-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-004-LDN' tries to include nonexistent context 'outrt-004-LDN-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-003-Celular044' tries to include nonexistent context 'outrt-003-Celular044-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-002-9_outside' tries to include nonexistent context 'outrt-002-9_outside-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outrt-001-01900' tries to include nonexistent context 'outrt-001-01900-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'outbound-allroutes' tries to include nonexistent context 'outbound-allroutes-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'from-did-direct-ivr' tries to include nonexistent context 'from-did-direct-ivr-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-local' tries to include nonexistent context 'ext-local-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-8' tries to include nonexistent context 'macro-from-zaptel-8-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-7' tries to include nonexistent context 'macro-from-zaptel-7-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-6' tries to include nonexistent context 'macro-from-zaptel-6-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-5' tries to include nonexistent context 'macro-from-zaptel-5-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-4' tries to include nonexistent context 'macro-from-zaptel-4-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-3' tries to include nonexistent context 'macro-from-zaptel-3-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-2' tries to include nonexistent context 'macro-from-zaptel-2-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'macro-from-zaptel-1' tries to include nonexistent context 'macro-from-zaptel-1-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-did-catchall' tries to include nonexistent context 'ext-did-catchall-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-did' tries to include nonexistent context 'ext-did-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-did-0002' tries to include nonexistent context 'ext-did-0002-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-did-0001' tries to include nonexistent context 'ext-did-0001-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'ext-test' tries to include nonexistent context 'ext-test-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'app-chanspy' tries to include nonexistent context 'app-chanspy-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'app-zapbarge' tries to include nonexistent context 'app-zapbarge-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'app-pickup' tries to include nonexistent context 'app-pickup-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: Context 'app-userlogonoff' tries to include nonexistent context 'app-userlogonoff-custom' [May 13 11:33:06] WARNING[24787]: pbx.c:6344 ast_context_verify_includes: == Manager 'admin' logged off from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found [May 13 11:33:10] NOTICE[3642]: chan_iax2.c:9186 __iax2_poke_noanswer: Peer '3072' is now UNREACHABLE! Time: 2 [May 13 11:33:20] NOTICE[3650]: chan_iax2.c:8264 socket_process: Peer '3072' is now REACHABLE! Time: 3 -- Starting simple switch on 'DAHDI/4-1' -- Executing [s@from-pstn:1] Set("DAHDI/4-1", "__FROM_DID=s") in new stack -- Executing [s@from-pstn:2] Gosub("DAHDI/4-1", "app-blacklist-check|s|1") in new stack -- Executing [s@app-blacklist-check:1] LookupBlacklist("DAHDI/4-1", "") in new stack -- Executing [s@app-blacklist-check:2] GotoIf("DAHDI/4-1", "0?blacklisted") in new stack -- Executing [s@app-blacklist-check:3] Return("DAHDI/4-1", "") in new stack -- Executing [s@from-pstn:3] ExecIf("DAHDI/4-1", "1 |Set|CALLERID(name)=3336060867") in new stack -- Executing [s@from-pstn:4] SetMusicOnHold("DAHDI/4-1", "setramex2") in new stack -- Executing [s@from-pstn:5] Set("DAHDI/4-1", "__MOHCLASS=setramex2") in new stack -- Executing [s@from-pstn:6] Set("DAHDI/4-1", "__CALLINGPRES_SV=allowed_not_screened") in new stack -- Executing [s@from-pstn:7] SetCallerPres("DAHDI/4-1", "allowed_not_screened") in new stack -- Executing [s@from-pstn:8] Goto("DAHDI/4-1", "timeconditions|1|1") in new stack -- Goto (timeconditions,1,1) -- Executing [1@timeconditions:1] GotoIfTime("DAHDI/4-1", "09:00-18:30|mon-fri|*|*?ivr-4|s|1") in new stack -- Goto (ivr-4,s,1) -- Executing [s@ivr-4:1] Set("DAHDI/4-1", "MSG=custom/setramex_conmutador_intro3") in new stack -- Executing [s@ivr-4:2] Set("DAHDI/4-1", "LOOPCOUNT=0") in new stack -- Executing [s@ivr-4:3] Set("DAHDI/4-1", "__DIR-CONTEXT=default") in new stack -- Executing [s@ivr-4:4] Set("DAHDI/4-1", "_IVR_CONTEXT_ivr-4=") in new stack -- Executing [s@ivr-4:5] Set("DAHDI/4-1", "_IVR_CONTEXT=ivr-4") in new stack -- Executing [s@ivr-4:6] GotoIf("DAHDI/4-1", "0?begin") in new stack -- Executing [s@ivr-4:7] Answer("DAHDI/4-1", "") in new stack -- Executing [s@ivr-4:8] Wait("DAHDI/4-1", "1") in new stack -- Executing [s@ivr-4:9] Set("DAHDI/4-1", "TIMEOUT(digit)=3") in new stack -- Digit timeout set to 3 -- Executing [s@ivr-4:10] Set("DAHDI/4-1", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 -- Executing [s@ivr-4:11] Set("DAHDI/4-1", "__IVR_RETVM=") in new stack -- Executing [s@ivr-4:12] ExecIf("DAHDI/4-1", "1|Background|custom/setramex_conmutador_intro3") in new stack [May 13 11:33:24] WARNING[24804]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 -- Playing 'custom/setramex_conmutador_intro3' (language 'en') -- Executing [0@ivr-4:1] DBdel("DAHDI/4-1", "") in new stack -- Executing [0@ivr-4:2] Set("DAHDI/4-1", "__NODEST=") in new stack -- Executing [0@ivr-4:3] Goto("DAHDI/4-1", "ext-group|6007|1") in new stack -- Goto (ext-group,6007,1) -- Executing [6007@ext-group:1] Macro("DAHDI/4-1", "user-callerid|") in new stack -- Executing [s@macro-user-callerid:1] Set("DAHDI/4-1", "AMPUSER=3336060867") in new stack -- Executing [s@macro-user-callerid:2] GotoIf("DAHDI/4-1", "0?report") in new stack -- Executing [s@macro-user-callerid:3] ExecIf("DAHDI/4-1", "1|Set|REALCALLERIDNUM=3336060867") in new stack -- Executing [s@macro-user-callerid:4] Set("DAHDI/4-1", "AMPUSER=") in new stack -- Executing [s@macro-user-callerid:5] Set("DAHDI/4-1", "AMPUSERCIDNAME=") in new stack -- Executing [s@macro-user-callerid:6] GotoIf("DAHDI/4-1", "1?report") in new stack -- Goto (macro-user-callerid,s,10) -- Executing [s@macro-user-callerid:10] GotoIf("DAHDI/4-1", "0?continue") in new stack -- Executing [s@macro-user-callerid:11] Set("DAHDI/4-1", "__TTL=64") in new stack -- Executing [s@macro-user-callerid:12] GotoIf("DAHDI/4-1", "1?continue") in new stack Mon, 13 May 2013 13:39:46 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121452-music-on-hold.html#121758 Hmmm it's weird. I'll try to test it later on my dumy sever. I'll let you know. Mon, 13 May 2013 13:21:33 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/121690-buzon-de-voz-central-de-oyentes-de-radio-disney.html#121757 Una vez que presionas el #, se guarda la grabación, ya no es necesario presionar 1 (haz la prueba para confirmarlo). Si es que quieres cambiar el audio, revisa en el CLI (asterisk -r) el nombre del audio que se ejecuta, luego, lo puedes reemplazar con otro o modificar el plan de marcado para colgar al instante. Mon, 13 May 2013 13:10:24 -0500 http://www.elastix.org/index.php/en/component/kunena/50-novatos/120956-sin-sonido-saliente.html#121751 que tal buen dia yo tuve el mismo problema, de la extensiones internas se escuchaba correctamento pero cuando llamaba hacia afuera no me escuchaban lo que hice fue poner un router marca DD-WRT, y este no se bien que funcion hace pero cambia todo el esquema. tengo lineas en 3 zonas y solo con ese router se me quitan todos los problemas. Mon, 13 May 2013 12:04:31 -0500 http://www.elastix.org/index.php/en/component/kunena/18-modules/120313-gabcast-channel.html#121750 kdacosta, It appears that the gabcast.com is no longer. I may be wrong and hope that someone here has more info on the subject. If they are no longer then maybe a few of us here need to set up something like it? I'm not sure I fully understand the reasoning behind gabcast but I'm up to learning more. amphibian Mon, 13 May 2013 11:51:57 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/121743-modificar-archivo-mastercsv.html#121743 estimados tengo un tarificador para el elastix pero tengo que modificar el stream del archivo master.csv. Este debe ser el stream que deve votar el elastix. "5430","3735251","2011-08-03 18:32:58","2011-08-03 18:33:15","37" como cambio o modifico los parametros de salida del stream. Mon, 13 May 2013 10:55:30 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/121701-smtp-authentication.html#121733 you will need to back off and put the original configs back until you find out what you did wrong. Mon, 13 May 2013 09:56:13 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/101419-guardar-las-grabaciones-en-otra-ruta.html?limit=10&start=10#121728 Hola, La idea del Script es la que se trata aqui: http://www.elastix.org/index.php/es/component/kunena/31-asterisk/107145-change-call-recording-names-and-sort-into-folders.html Básicamente seria el mismo script pero apuntando a una unidad de red de tu NAS que montas previamente en el filesystem del Elastix. Por otra parte, puede que no requieras cambiar nada del modulo de monitoring, ya que usando el script lo que generas en el servidor NAS, es una copia a tiempo real. Pero al quedar las mismas grabaciones bajo la ruta original (/var/spool/asterisk/monitor), la pestaña de monitoring te funcionará sin problemas. Así que si solo quieres el tema del respaldo, el script es ya suficiente para tí. Saludos, Claudio Mon, 13 May 2013 08:22:17 -0500 http://www.elastix.org/index.php/en/component/kunena/116-security/113383-fake-auth-rejection-for-device-5550000.html#121723 Here is the only solution I know for now is to use an alternative port for SIP accounts from random IPs. On my firewall I have a list of known IPs that are allowed for port 5060. For everyone else they are are available only if they use for example port 5062. After I setup all outside SIP clients to use port 5062 I turned off 5060 at the firewall and suddenly all of the hackers stopped. Last night I enabled 5060 and within a short time the hackers were back. Considering that IPs show up as the local IP I have no other way to stop this other than use an alternative IP. Mon, 13 May 2013 07:16:15 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/118632-strange-line-on-my-full-log.html#121722 Here is the only solution I know for now is to use an alternative port for SIP accounts from random IPs. On my firewall I have a list of known IPs that are allowed for port 5060. For everyone else they are av available only if they use for example port 5062. After I setup all outside SIP clients to use port 5062 I turned off 5060 at the firewall and suddenly all of the hackers stopped. Last night I enabled 5060 and within a short time the hackers were back. Considering that IPs show up as the local IP I have no other way to stop this other than use an alternative IP. Mon, 13 May 2013 07:09:57 -0500 http://www.elastix.org/index.php/en/component/kunena/84-le-coin-du-debutant/121718-numero-dappel-de-call-center.html#121718 bonjour a vous j'ai mis sur pied un call center et un trunk IAX2 entre deux serveurs elaxtix 2.3 de numerotation 201 et 4001 et un ring group qand le 4001 appel le 201 l'appel pass bien mais dès lors que le 201 ce met en mode agent call center le 4001 appel le numero de groupe 200 ou directement le 201 le svi ne prend plus on dit call busy here svp aidezz moi a permettre au 4001 d'appeler le 201 lorsqu'il est en mode call center et s'il ya un prefixe d'appel call center a configurer dit moi comment faire merci pour vos reponses Mon, 13 May 2013 06:01:37 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121710-b410p-and-caller-id.html#121717 tomov, Check whether your carrier has it turned on or not first.... In the Oceanic region, (it may be the same in the rest of the world) it is common that this is charged as an extra service on BRI and PRI lines (sometimes that lump it in with the word "Extension Level Charging", which is a crock, as most do not want extension level charging, but this is the only way to get the Caller ID turned on. Regards Bob Mon, 13 May 2013 05:57:27 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121613-no-service.html?limit=10&start=10#121716 If you run into issues understanding, then post back...many can assist. But in particular, you want to turn it on when you get the phone working, so we can see a normal SIP transmission. But leave it on, so we can see what is transmitted when the phone disconnects...this is the important part, and if you can post the relevant part (not the whole thing), it might give us an idea what is wrong... Regards Bob Mon, 13 May 2013 05:54:07 -0500 http://www.elastix.org/index.php/en/component/kunena/116-security/121696-security-and-flash-operator-panel.html#121714 Make sure that you have the ports open : TCP/4445 (the normal FOP) TCP/4446 (sometimes if you install FOP2) But this is if you are using flash operator panel If you are using the Elastix Operator Panel, then I think this is covered. Also if this is not your issue, then try another browser and update your flash...this was a common issue in the early days... Regards Bob Mon, 13 May 2013 05:42:00 -0500 http://www.elastix.org/index.php/en/component/kunena/25-newbies-corner-/116669-cant-login.html?limit=10&start=20#121713 Check if your disk isn't full. You can delete old backups from /var/lib/asterisk/backups or from /var/www/backups or some monitoring files. Then try login again. Mon, 13 May 2013 05:34:29 -0500 http://www.elastix.org/index.php/en/component/kunena/11-ayuda/121689-cambio-de-https-por-http.html#121691 Hola, Fijate aqui: http://www.elastix.org/index.php/es/component/kunena/116-security/112379-how-to-change-default-ports-of-http-a-https-.html No es directamente lo que quieres, pero ya te darás cuenta que cambiar desde ahi... Saludos, Claudio Sun, 12 May 2013 19:56:43 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121686-connect-intercom-with-elastix--help-me.html#121687 Note : I'm not that expert in Elastix :) Sun, 12 May 2013 16:25:53 -0500 http://www.elastix.org/index.php/en/component/kunena/72-otros/121682-grandstream-ht-503-v11b-llamadas-entrantes.html#121682 Buenos Dias, estoy empezando a dar mi primeros pasos, Situacion tengo el dispositivo registrado segun la visualizacion observada via web tanto para el puerto fxo como fxs, y puedo realizar llamadas salientes perfectamente. El tema son las llamadas entrantes, que el dispositivo las detecta y la deriva a la extension correspondiente esto lo observo via la consola, pero mi dolor de cabeza y sobre el cual estoy hace 2 dias, es que no me reproduce ningun, sonido, en la consolo llego a observar el mensaje de despedida, y obviamento no lo escucho y hasta cierra la comunicacion Lo mas gracioso, es que si yo llamo a la extension desde un interno la escucho perfectamente, como asi tambien si lo llamo a una entrante DID, que configure y escucho tambien el mensaje Alguien tiene alguna idea, para orientarme Desde ya muchas gracias, SALIDA DE LA CONSOLA con la situacion comentada -- Goto (day-menu,s,1) -- Executing [s@day-menu:1] Answer("SIP/ata-fxo-00000116", "") in new stack -- Executing [s@day-menu:2] Wait("SIP/ata-fxo-00000116", "0.5") in new stack -- SIP/troncalVOIPMS-00000115 answered SIP/200-00000114 -- Executing [s@day-menu:3] BackGround("SIP/ata-fxo-00000116", "tt-monkeys") in new stack -- Playing 'tt-monkeys.alaw' (language 'es') == Spawn extension (users, 46375627, 1) exited non-zero on 'SIP/200-00000114' -- Executing [s@day-menu:4] WaitExten("SIP/ata-fxo-00000116", "5") in new stack -- Timeout on SIP/ata-fxo-00000116, going to 't' -- Executing [t@day-menu:1] Playback("SIP/ata-fxo-00000116", "goodbye") in new stack -- Playing 'goodbye.gsm' (language 'es') -- Executing [t@day-menu:2] Hangup("SIP/ata-fxo-00000116", "") in new stack == Spawn extension (day-menu, t, 2) exited non-zero on 'SIP/ata-fxo-00000116' Archivo sip [general] context=noasignado allowguest=no srvlookup=no udpbindaddr=0.0.0.0 tcpenable=no disallow=all allow=alaw language=es [ata-fxo] type=friend host=dynamic secret=S3cr3t0 context=features mailbox=400@default qualify=yes Configuraciones HT 503 STATUS Product Model: HT-503 V1.1B Software Version: Program-- 1.0.5.5 Bootloader-- 1.0.0.15 Core-- 1.0.5.2 Base-- 1.0.5.2 System Up Time: 09:46:05 up 1:06 PPPoE Link Up: Disabled NAT: Port Status: Port Hook Registration DND Forward Busy Forward Delayed Forward FXS On Hook Registered No FXO Idle Registered No PUERTO FXO Preferred DTMF method: 1 Prioridad RFC2833 Preferred Vocoder: (in listed order) choice 1: pcmu choice 2: pcma choice 3: 723 choice 4: 729 choice 5: 726-32 choice 6: ilbc choice 7: g729e choice 8: aal2 PUERTO FXS unregister on reboot yes Prefered DTMF method Priority 1 RFC2833 uncondicional call forward to VOIP 507 67.22.XXXX 5060 DE NUEVO MUCHAS GRACIAS, Sun, 12 May 2013 10:13:50 -0500 http://www.elastix.org/index.php/en/component/kunena/7-old-forum/7982-hardware-detection-does-not-get-the-sangoma-a500.html?limit=10&start=10#121677 I understand this question. Let's discuss. Sun, 12 May 2013 06:52:06 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/121601-how-to-handle-two-different-companies-on-handsets.html#121676 hinzinho, Thanks for posting back.....appreciate it. Was just having a think why this was the case (especially as phone transfers will often lose the CID)....but just realised, this is a pure Asterisk function, which is why the CID is preserved. Regards Bob Sun, 12 May 2013 06:50:48 -0500 http://www.elastix.org/index.php/en/component/kunena/31-asterisk/121675-problem-with-remote-connection.html#121675 Hello, i have elastix with zycoo zx100 pbx i set internet ip for wan and set local ip for lan and my local staff connect to lan port and some of my remote staff connect to wan port via internet and also in elastix i enable nat (enable it in sip_nat.conf and do prerouting with iptables) but i have a problem after some hours when my remote staff call everywhere his voice sent and the receiver can hear his voice but the when caller receiver talk my staff does not hear back. do you have any idea for solve this issue? and when i reboot my device everthing back to normal and works perfect, thanks lot, Sun, 12 May 2013 06:30:50 -0500 http://www.elastix.org/index.php/en/component/kunena/3-help/101338-faxing-via-microsoft-outlook.html#121673 hey i know this is a old post but did you every got this working and if you did can you sure some info please Sun, 12 May 2013 03:13:32 -0500