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  1. itjumper

    remote extension via 3g

    I have a sip trunk from vbuzzer attached to my elastix box. When I connect an extension via wifi to my elastix box, incoming and going calls are good. However, when I connect the same extension via 3g (I am using a Nokia N95), there is no sound at both ends. The only difference in my...
  2. itjumper

    Incoming call assigned to incorrect sip trunk

    Using Elastix 1.6-12. I have 2 accounts with CIK, 111111 and 222222. When call coiming in from 111111, it always assigns it to peer 222222. This is normally not a problem, but if I initate a sip REFER, then the incorrect username/password is sent to CIK and the transfer will fail. The...
  3. itjumper

    Clean up old FAX

    How can I clean up (remove) old faxes from elastix ? I have deleted some faxes in Fax Visor, but the tif files are still there. I would like to run a scheduled job to remove fax older than a certain age from Fax Visor and /var/spool/hylafax/recvq Thanks
  4. itjumper

    using asterisk as an extension

    I have some accounts to an provider using analog adaptor (SPA2102). I can do a blind transfer by hitting "flash", *98<destination> on an analog phone (*98 is the blind transfer feature code on spa2102). SPA2102 will send something to my provider and initiates the transfer. I can still use my...
  5. itjumper

    FAX not working (incoming)

    Using Elastix 1.5.2 and same result with Digium TDM410 and Openvox A1200. Fax is not received, asterisk answered and detected FAX. 1. Set up IAX extension 10001, with email address 2. Set FAX extension on general settings to disabled 2. Set up inbound route FAX extension to 10001, zaptel, 2...
  6. itjumper

    Got SIP response 489 "Bad event"

    I'm trying to use an sip extension created at a remote site as a trunk here in our office. Outgoing calls work fine, but could not get incoming call. The extension is showed as registered on the remote site. We keep getting "Got SIP response 489 "Bad event" back from ..." in the log of...
  7. itjumper

    Avaya h323

    Can any one show me doco on how to install h323 in Elastix ??
  8. itjumper

    FastAGI

    anyone has any idea how to set up an agi server ??
  9. itjumper

    AGI / FastAGI

    I have a test agi in php and it works ok when ran as an agi with the following call from the dialplan exten => 1234,1,AGI(t.php) I would like to centralize all scripts to a agi server. To start, I change the call to exten => 1234,1,AGI(agi://localhost/t.php) so that it still runs...
  10. itjumper

    CIKTel account

    I have an account from CIK (www.CIKTel.com). This account is supposed to be an extension in CIK system, i.e. using SIP phone. I want to config it as a trunk in asterisk. This is my trunk setting canreinvite=no context=from-trunk fromdomain=CMTor1.ciktel.com fromuser=xxxxxxx...
  11. itjumper

    clamav, mailscanner ...

    Elastix comes with an email server. I would like to add clamav, mailscanner, spamassissin tot it. Anyone knows how to do it?
  12. itjumper

    follow me, ring group no answer not working

    Using Elastix 1.5.2. I have set up destination if no answer to a voicemail. If there is only one entry on the list (follow-me or ring group), everthing works fine. But if there are more on the list, then the voicemail will not be called. This might be a bug.
  13. itjumper

    VoIP Provider call forward

    I have a couple of VoIP lines connecting to my elastix server via PAP2 / SPA3102. VoIP <--> PAP2 <--> SPA3102 <--> elastix One of my VoIP accounts provides unlimited North America calling, including call forwarding, with a local phone # (416-111-1111). In order to reduce the occupied time...
  14. itjumper

    Nortel i2002 pgone

    Have anyone used this phone with elastix. I know UNISTIM needs to be added to the kernel. Anyone tried ??
  15. itjumper

    Elastix PXE install

    Anyone knows if Elastix can be installed from a PXE server? And if yes, how to set up TFTP and PXE Any help appreciated.
  16. itjumper

    Live backup

    Anyone knows how to backup a live Elastix? I have tried dd and partimage. Both failed to backup a LIVE partition. I tried to install mondo, but lots of dependencies are required. I would appreciate if anyone has a detailed instruction on backing up a LIVE partition, w.r.t. Elastix 1.3
  17. itjumper

    Calling card - success

    Early this year a friend of mine asked me if we could start a business with asterisk. Since I knew nothing about asterisk, and PBX in general, I did a lot of research. I found Trixbox, PIAF and Elastix. I tried them all and finally settled on Elastix 0.98. With my 20 years of programming...
  18. itjumper

    Elastix 1.3 - yum deos not work

    Tried to install nano via yum, but could not find http://repo.elastix.org/centos/5/update ... repomd.xml Interent connection was fine [root@elastix ~]# ping www.yahoo.com PING www.yahoo.com (69.147.76.15) 56(84) bytes of data. 64 bytes from f1.www.vip.re1.yahoo.com (69.147.76.15)...
  19. itjumper

    Elastix 1.3-2 - asterisk failed to start

    Did a fresh install and asterisk failed to start. retrieve_conf failed, config not applied Reload failed because retrieve_conf encountered an error: 1 Added ago (freepbx.RCONFFAIL) Asterisk Manager Connection Failure Failed to connect to the Asterisk manager through port: 5038 Added...
  20. itjumper

    Feature code with Lisksys phones

    I use Grandstream, Polycom and LinkSys IP phones. When I dial from Grandstream and Polycom phones, intercom works fine. For instance, when I dial *801001, extension 1001 will auto answer with speaker on. I can see *801001 being dialed in asterisk console. But when I dial *801001 from a...
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