ht 503 fxo gataway

olivier

Joined
Sep 13, 2013
Messages
28
Points
0
desole

Code: 
ontent-Length: 241

v=0
o=root 1014505747 1014505747 IN IP4 x.x.x
s=Asterisk PBX 1.8.20.0
c=IN IP4 78.240.x.x
t=0 0
m=audio 16038 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/testpsnt/298319530
Really destroying SIP dialog 'MTBmZDVhOTQxZGYwOTA3Mzg5MGE5OTdlM2M1MWRmZWY.' Method: REGISTER

<--- SIP read from UDP:192.168.1.61:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 78.240.x.x:5060;branch=z9hG4bK36a842f7;rport=5060;received=192.168.1.38
From: "tete" <sip:103@192.168.1.61>;tag=as20155f3e
To: <sip:298319530@192.168.1.61:5062>
Call-ID: 48f03a9078f2df4a2124af5735e94549@192.168.1.61
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.9.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.61:5062 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 78.240.x.x:5060;branch=z9hG4bK36a842f7;rport=5060;received=192.168.1.38
From: "tete" <sip:103@192.168.1.61>;tag=as20155f3e
To: <sip:298319530@192.168.1.61:5062>;tag=1556268859
Call-ID: 48f03a9078f2df4a2124af5735e94549@192.168.1.61
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.9.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 192.168.1.61:5062:
ACK sip:298319530@192.168.1.61:5062 SIP/2.0
Via: SIP/2.0/UDP 78.240.x.x:5060;branch=z9hG4bK36a842f7;rport
Max-Forwards: 70
From: "tete" <sip:103@192.168.1.61>;tag=as20155f3e
To: <sip:298319530@192.168.1.61:5062>;tag=1556268859
Contact: <sip:103@78.240.x.x:5060>
Call-ID: 48f03a9078f2df4a2124af5735e94549@192.168.1.61
CSeq: 102 ACK
User-Agent: FPBX-2.8.1(1.8.20.0)
Content-Length: 0


---
y a plus mais ca se repete

merci
 
danardf

danardf

Joined
Dec 3, 2007
Messages
8,069
Points
88
Ton réseau LAN est dans quel subnet?
Parce que j'ai un tas d'inconnues comme.
C'est quoi çà : 78.x.x.x ????
192.168.1.61 Donc.... ok c'est ton HT503

Ce qui veut dire que ton réseau doit être 192.168.1.0/24
Donc...
Ton serveur Elastix 192.168.1.38/24
Ta passerelle HT503 192.168.1.61/24
Ton routeur par défaut 192.168.1.1/24

Comment est branché ton HT503?
C'est sur ta patte WAN ou LAN?
 

olivier

Joined
Sep 13, 2013
Messages
28
Points
0
re,
lan 192.168.1.x/24
rj54 sur lan ht 503 config ht 503 :192.168.1.61/24
passerelle:192.168.1.254(freeboule)
elastix :192.168.1.38
 
danardf

danardf

Joined
Dec 3, 2007
Messages
8,069
Points
88
Ouai ... tu n'as pas pris en considération que j'avais branché le HT503 sur la patte WAN?
Je l'ai dit en haut de cette page:
....
Par contre, Je n'ai qu'une RJ45 de mon LAN branché sur mon routeur (switch 4 ports).
La RJ est branchée sur le port WAN et non sur le port LAN
Donc, pas de DHCP, et pas de nat. (ça ne sert à rien)
.....
Click to expand...
 

olivier

Joined
Sep 13, 2013
Messages
28
Points
0
re

l habitude tue, automatisme . :)
Code: 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="746ceedd"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2127364203-5060-41@BJC.BGI.B.EG' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.46:5060 --->
ACK sip:0298319530@192.168.1.38 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK1927354973;rport
From: "psnt" <sip:103@192.168.1.38>;tag=798814868
To: <sip:0298319530@192.168.1.38>;tag=as039934bd
Call-ID: 2127364203-5060-41@BJC.BGI.B.EG
CSeq: 400 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.46:5060 --->
INVITE sip:0298319530@192.168.1.38 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK1673250477;rport
From: "psnt" <sip:103@192.168.1.38>;tag=798814868
To: <sip:0298319530@192.168.1.38>
Call-ID: 2127364203-5060-41@BJC.BGI.B.EG
CSeq: 401 INVITE
Contact: "psnt" <sip:103@192.168.1.46:5060>
Authorization: Digest username="103", realm="asterisk", nonce="746ceedd", uri="sip:0298319530@192.168.1.38", response="679a82d10349596a8415846b864bbcd3", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP1165 1.0.4.14
Privacy: none
P-Preferred-Identity: "psnt" <sip:103@192.168.1.38>
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 401

v=0
o=103 8000 8000 IN IP4 192.168.1.46
s=SIP Call
c=IN IP4 192.168.1.46
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 19 lines) ---
Sending to 192.168.1.46:5060 (NAT)
Using INVITE request as basis request - 2127364203-5060-41@BJC.BGI.B.EG
Found peer '103' for '103' from 192.168.1.46:5060
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x1d0d (g723|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.46:5004
Looking for 0298319530 in from-internal (domain 192.168.1.38)
list_route: hop: <sip:103@192.168.1.46:5060>

<--- Transmitting (NAT) to 192.168.1.46:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK1673250477;received=192.168.1.46;rport=5060
From: "psnt" <sip:103@192.168.1.38>;tag=798814868
To: <sip:0298319530@192.168.1.38>
Call-ID: 2127364203-5060-41@BJC.BGI.B.EG
CSeq: 401 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0298319530@78.x.x.x:5060>
Content-Length: 0


<------------>
    -- Executing [0298319530@from-internal:1] Macro("SIP/103-00000006", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/103-00000006", "AMPUSER=103") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/103-00000006", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/103-00000006", "1?Set(REALCALLERIDNUM=103)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/103-00000006", "AMPUSER=103") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/103-00000006", "AMPUSERCIDNAME=tete") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/103-00000006", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/103-00000006", "AMPUSERCID=103") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/103-00000006", "CALLERID(all)="tete" <103>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/103-00000006", "1?Set(CHANNEL(language)=fr)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/103-00000006", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/103-00000006", "CALLERID(number)=103") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/103-00000006", "CALLERID(name)=tete") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/103-00000006", "Using CallerID "tete" <103>") in new stack
    -- Executing [0298319530@from-internal:2] NoOp("SIP/103-00000006", "Calling Out Route: psnt") in new stack
    -- Executing [0298319530@from-internal:3] Set("SIP/103-00000006", "MOHCLASS=default") in new stack
    -- Executing [0298319530@from-internal:4] Set("SIP/103-00000006", "_NODEST=") in new stack
    -- Executing [0298319530@from-internal:5] Macro("SIP/103-00000006", "record-enable,103,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/103-00000006", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/103-00000006", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/103-00000006", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/103-00000006", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/103-00000006", "1?MacroExit()") in new stack
    -- Executing [0298319530@from-internal:6] Macro("SIP/103-00000006", "dialout-trunk,2,298319530,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/103-00000006", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/103-00000006", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/103-00000006", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/103-00000006", "DIAL_NUMBER=298319530") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/103-00000006", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/103-00000006", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/103-00000006", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/103-00000006", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/103-00000006", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/103-00000006", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/103-00000006", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/103-00000006", "0?Set(REALCALLERIDNUM=103)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/103-00000006", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/103-00000006", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/103-00000006", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/103-00000006", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/103-00000006", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/103-00000006", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/103-00000006", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/103-00000006", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/103-00000006", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/103-00000006", "1?sub-flp-2,s,1") in new stack
    -- Executing [s@sub-flp-2:1] ExecIf("SIP/103-00000006", "1?Return()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/103-00000006", "OUTNUM=298319530") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/103-00000006", "custom=SIP/testpsnt") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/103-00000006", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/103-00000006", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/103-00000006", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/103-00000006", "0?bypass
j essaye de comprendre les log mais bon, la c'est du coreen

merci
 

olivier

Joined
Sep 13, 2013
Messages
28
Points
0
re,

il y a aussi une chose qui megene , cest sur le ht 503 fxo reste idle comme si rien n arrivait a lui, alors que quand j appel avec mon tel classique , il est busy le fx0.

c'est comme si elastix n avait pas la bonne route.

merci
 

olivier

Joined
Sep 13, 2013
Messages
28
Points
0
j ai aussi ca comme erreur
je l ai trouver dans freepbx asterix sip setting.

je cherche je cherche mais je sais pas ou je vais ..dans le mur :lol:
Code: 
[b]ERRORS

    Settings in /etc/asterisk/sip_general_custom.conf may override these. Those settings should be removed.[/b]

NAT Settings
NAT 	
yes 	no 	never 	route
IP Configuration 	
Public IP Static IP Dynamic IP
External IP 'mon ip fixe public free	
Local Networks 	/
	/
	
Audio Codecs
 
danardf

danardf

Joined
Dec 3, 2007
Messages
8,069
Points
88
Tu arrives des fois à te concentrer sur quelque chose? :lol:

Allez bonne soirée. :wink:
Je fais un break, et demain je travail sur un projet en retard.
Donc.
En attendent reprends le problème au début.
C'est tout ce que je peux te dire.
Si tu ne connais pas grand chose sur Elastix (Asterisk), prends le temps de lire des docs téléchargeables sur le site Elastix.
ça te sera toujours utile.
 

olivier

Joined
Sep 13, 2013
Messages
28
Points
0
merci

oui j arrive a me concentrer sur quelque chose :wink:

je n aime quand ca marche pas, en plus je suis plus sur les serveurs que sur la voip , (ca se voit).
Par contre la c 'est vrai je me suis eparpiller, en plus pour corser l affaire j ai monter ca sur un boitier alix, donc grosse prise de tete deja a la base.

j ai tout reinstaller ce soirn configue a zero.

Merci pour ton aide et patience, bon courage pour demain sur ton projet..

a+
 

olivier

Joined
Sep 13, 2013
Messages
28
Points
0
bonsoir,

j espere que tout c est bien passer pour ton projey.

moi j ai tout repris, canfig ht et elastix.
j ai imprimer tout document et verifier ligne par ligne le ht 503 (pas a l abri d une erreur quand meme) :)

Maintenant je n ai plus le message d erreur me disant que le serveur est occuper, j ai un pauvre ring et finex:

c'est un mieux , ca prouve que j ai avancer ou regresse,:)
Code: 
[root@localhost ~]# asterisk -r
Verbosity is at least 3
    -- Remote UNIX connection
localhost*CLI> sip set debug ip 192.168.1.38
SIP Debugging Enabled for IP: 192.168.1.38
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [0298319530@from-internal:1] Macro("SIP/103-00000030", "user-callerid,SKIPTTL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/103-00000030", "AMPUSER=103") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/103-00000030", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/103-00000030", "1?Set(REALCALLERIDNUM=103)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/103-00000030", "AMPUSER=103") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/103-00000030", "AMPUSERCIDNAME=tete") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/103-00000030", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/103-00000030", "AMPUSERCID=103") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/103-00000030", "CALLERID(all)="tete" <103>") in new stack
    -- Executing [s@macro-user-callerid:9] ExecIf("SIP/103-00000030", "1?Set(CHANNEL(language)=fr)") in new stack
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/103-00000030", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] Set("SIP/103-00000030", "CALLERID(number)=103") in new stack
    -- Executing [s@macro-user-callerid:20] Set("SIP/103-00000030", "CALLERID(name)=tete") in new stack
    -- Executing [s@macro-user-callerid:21] NoOp("SIP/103-00000030", "Using CallerID "tete" <103>") in new stack
    -- Executing [0298319530@from-internal:2] NoOp("SIP/103-00000030", "Calling Out Route: free_out") in new stack
    -- Executing [0298319530@from-internal:3] Set("SIP/103-00000030", "MOHCLASS=default") in new stack
    -- Executing [0298319530@from-internal:4] Set("SIP/103-00000030", "_NODEST=") in new stack
    -- Executing [0298319530@from-internal:5] Macro("SIP/103-00000030", "record-enable,103,OUT,") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/103-00000030", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/103-00000030", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/103-00000030", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,15)
    -- Executing [s@macro-record-enable:15] GotoIf("SIP/103-00000030", "0?IN") in new stack
    -- Executing [s@macro-record-enable:16] ExecIf("SIP/103-00000030", "1?MacroExit()") in new stack
    -- Executing [0298319530@from-internal:6] Macro("SIP/103-00000030", "dialout-trunk,2,0298319530,") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/103-00000030", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/103-00000030", "0?sub-pincheck,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/103-00000030", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/103-00000030", "DIAL_NUMBER=0298319530") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/103-00000030", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/103-00000030", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/103-00000030", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/103-00000030", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/103-00000030", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/103-00000030", "outbound-callerid,2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/103-00000030", "0?Set(CALLERPRES()=)") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/103-00000030", "0?Set(REALCALLERIDNUM=103)") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/103-00000030", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/103-00000030", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/103-00000030", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/103-00000030", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/103-00000030", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/103-00000030", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/103-00000030", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/103-00000030", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/103-00000030", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
    -- Executing [s@macro-dialout-trunk:12] GosubIf("SIP/103-00000030", "1?sub-flp-2,s,1") in new stack
    -- Executing [s@sub-flp-2:1] ExecIf("SIP/103-00000030", "1?Return()") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/103-00000030", "OUTNUM=0298319530") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/103-00000030", "custom=SIP/psnt") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/103-00000030", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/103-00000030", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/103-00000030", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/103-00000030", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/103-00000030", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/103-00000030", "SIP/psnt/0298319530,300,") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/psnt/0298319530
    -- SIP/psnt-00000031 is ringing
    -- SIP/psnt-00000031 answered SIP/103-00000030
    -- Locally bridging SIP/103-00000030 and SIP/psnt-00000031
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/103-00000030", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/103-00000030", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/103-00000030", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/103-00000030", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/103-00000030", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/103-00000030", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/103-00000030", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/103-00000030", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/103-00000030", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/103-00000030", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/103-00000030", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/103-00000030", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/103-00000030", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/103-00000030>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/103-00000030", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/103-00000030' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/103-00000030'
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/103-00000030' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 0298319530, 6) exited non-zero on 'SIP/103-00000030'
localhost*CLI>
j espere que ce bout de code te vas

merci
a+
 
danardf

danardf

Joined
Dec 3, 2007
Messages
8,069
Points
88
Salut.

Mon projet n'avance pas trop comme je veux mais il y a du mieux. J'ai plus de billes pour travailler.
Mais j'ai encore du taf.

Comme çà déjà je ne sais pas pourquoi tu valides un sip debug et que tu n'aies pas de de trace SIP dans ta capture!
Pourquoi lancer Asterisk sans verbosity ?: asterisk -r , mettre asterisk -rvvvvvvvvvv

A la louche, ton appel depuis le 103 part bien sur la passerelle pour composer le 02...etc
Je pense qu'il doit y avoir un problème de conf sur ta passerelle filtrant ton n° émis.
Je pense au paramètre dial plan dans le FXO entre autre. {0xxxxxxxxx | 3xxx | 1x | 11x}

Vérifies encore et encore.
Il n'y a que toi qui puisse te dépanner tu as le serveur et la passerelle à tes côtés. :wink:
Mais à mon avis, tu as due passer à côté d'un truc.

Allez je te laisse. Je vais bosser un peu.
 

olivier

Joined
Sep 13, 2013
Messages
28
Points
0
salut

j ai fais un sip set debug on car je n avais pas plus d infos avec sip set debug ip 192.168.1.38.
Comme je ne sais pas a quoi resemble une trame debug.
Code: 
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP monipfree:5060;branch=z9hG4bK753b2496;rport=5060;received=192.168.1.38
From: "Unknown" <sip:Unknown@monipfree>;tag=as2e8a78fc
To: <sip:103@192.168.1.46:5060>;tag=380443998
Call-ID: 5a644c7471c0b4553ac1597207380174@monipfree:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXP1165 1.0.4.14
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5a644c7471c0b4553ac1597207380174@monipfree:5060' Method: OPTIONS
Retransmitting #6 (NAT) to 192.168.1.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK825779616;received=192.168.1.46;rport=5060
From: "psnt" <sip:103@192.168.1.38>;tag=626545817
To: <sip:0298319530@192.168.1.38>;tag=as3bcee7e8
Call-ID: 2095547050-5060-6@BJC.BGI.B.EG
CSeq: 51 INVITE
Server: FPBX-2.8.1(1.8.20.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:0298319530@monipfree:5060>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1700353902 1700353902 IN IP4 monipfree
s=Asterisk PBX 1.8.20.0
c=IN IP4 monipfree
t=0 0
m=audio 18976 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Executing [h@macro-dialout-trunk:1] Macro("SIP/103-00000008", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/103-00000008", "1?endmixmoncheck") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] NoOp("SIP/103-00000008", "End of MIXMON check") in new stack
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/103-00000008", "1?nomeetmemon") in new stack
    -- Goto (macro-hangupcall,s,28)
    -- Executing [s@macro-hangupcall:28] NoOp("SIP/103-00000008", "End of MEETME check") in new stack
    -- Executing [s@macro-hangupcall:29] GotoIf("SIP/103-00000008", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,34)
    -- Executing [s@macro-hangupcall:34] NoOp("SIP/103-00000008", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:35] GotoIf("SIP/103-00000008", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,41)
    -- Executing [s@macro-hangupcall:41] NoOp("SIP/103-00000008", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:42] GotoIf("SIP/103-00000008", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,45)
    -- Executing [s@macro-hangupcall:45] GotoIf("SIP/103-00000008", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,48)
    -- Executing [s@macro-hangupcall:48] GotoIf("SIP/103-00000008", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,50)
    -- Executing [s@macro-hangupcall:50] AGI("SIP/103-00000008", "hangup.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/hangup.agi
    -- <SIP/103-00000008>AGI Script hangup.agi completed, returning 0
    -- Executing [s@macro-hangupcall:51] Hangup("SIP/103-00000008", "") in new stack
  == Spawn extension (macro-hangupcall, s, 51) exited non-zero on 'SIP/103-00000008' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/103-00000008'
Scheduling destruction of SIP dialog '5330b60c4e058974148ef99d59928a6a@192.168.1.61' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:201@192.168.1.61:5062> for address/port to send to
set_destination: set destination to 192.168.1.61:5062
Reliably Transmitting (NAT) to 192.168.1.61:5062:
BYE sip:201@192.168.1.61:5062 SIP/2.0
Via: SIP/2.0/UDP monipfree:5060;branch=z9hG4bK25df6aff;rport
Max-Forwards: 70
From: "tete" <sip:103@192.168.1.61>;tag=as6ee9bbf1
To: <sip:0298319530@192.168.1.61:5062>;tag=878524101
Call-ID: 5330b60c4e058974148ef99d59928a6a@192.168.1.61
CSeq: 103 BYE
User-Agent: FPBX-2.8.1(1.8.20.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/103-00000008' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 0298319530, 6) exited non-zero on 'SIP/103-00000008'
Scheduling destruction of SIP dialog '2095547050-5060-6@BJC.BGI.B.EG' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:103@192.168.1.46:5060> for address/port to send to
set_destination: set destination to 192.168.1.46:5060
Reliably Transmitting (NAT) to 192.168.1.46:5060:
BYE sip:103@192.168.1.46:5060 SIP/2.0
Via: SIP/2.0/UDP monipfree:5060;branch=z9hG4bK02a4ee97;rport
Max-Forwards: 70
From: <sip:0298319530@192.168.1.38>;tag=as3bcee7e8
To: "psnt" <sip:103@192.168.1.38>;tag=626545817
Call-ID: 2095547050-5060-6@BJC.BGI.B.EG
CSeq: 102 BYE
User-Agent: FPBX-2.8.1(1.8.20.0)
Proxy-Authorization: Digest username="103", realm="asterisk", algorithm=MD5, uri="sip:192.168.1.38", nonce="", response="50e28891d7911b4ba296f13edd9996cb"
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.61:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP monipfree:5060;branch=z9hG4bK25df6aff;rport=5060;received=192.168.1.38
From: "tete" <sip:103@192.168.1.61>;tag=as6ee9bbf1
To: <sip:0298319530@192.168.1.61:5062>;tag=878524101
Call-ID: 5330b60c4e058974148ef99d59928a6a@192.168.1.61
CSeq: 103 BYE
Contact: <sip:201@192.168.1.61:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.9.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '5330b60c4e058974148ef99d59928a6a@192.168.1.61' Method: INVITE

<--- SIP read from UDP:192.168.1.46:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP monipfree:5060;branch=z9hG4bK02a4ee97;rport=5060;received=192.168.1.38
From: <sip:0298319530@192.168.1.38>;tag=as3bcee7e8
To: "psnt" <sip:103@192.168.1.38>;tag=626545817
Call-ID: 2095547050-5060-6@BJC.BGI.B.EG
CSeq: 102 BYE
Contact: <sip:103@192.168.1.46:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1165 1.0.4.14
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '2095547050-5060-6@BJC.BGI.B.EG' Method: INVITE

<--- SIP read from UDP:192.168.1.37:56352 --->


<------------->
Reliably Transmitting (NAT) to 192.168.1.61:5062:
OPTIONS sip:192.168.1.61 SIP/2.0
Via: SIP/2.0/UDP monipfree:5060;branch=z9hG4bK5e200339;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@monipfree>;tag=as5c4a1170
To: <sip:192.168.1.61>
Contact: <sip:Unknown@monipfree:5060>
Call-ID: 0f9241891150f23b246506b324ef412f@monipfree:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 18 Jun 2014 19:09:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.61:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP monipfree:5060;branch=z9hG4bK5e200339;rport=5060;received=192.168.1.38
From: "Unknown" <sip:Unknown@monipfree>;tag=as5c4a1170
To: <sip:192.168.1.61>;tag=1557894736
Call-ID: 0f9241891150f23b246506b324ef412f@monipfree:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.9.1 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '0f9241891150f23b246506b324ef412f@monipfree:5060' Method: OPTIONS
localhost*CLI> sip set debug off
SIP Debugging Disabled
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
localhost*CLI>
La je reverifie les paramettres fxo du ht 503
merci
 
danardf

danardf

Joined
Dec 3, 2007
Messages
8,069
Points
88
Fais moi un schéma de ton installation (détaillée).

Comment est-ce possible de voir ton adresse IP Free dans la trace SIP?
Si tu téléphones depuis ton LAN, tu n'as aucune raison de voir ton adresse IP Free.

Code: 
Audio is at IP_Elastix port 10012
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to IP_HT503:5062:
INVITE sip:0172xxxxxx@IP_HT503:5062 SIP/2.0
Via: SIP/2.0/UDP IP_Elastix:5060;branch=z9hG4bK7cd7cc1c;rport
From: "0285xxxxxx" <sip:0285xxxxxx@IP_HT503>;tag=as1df6f147
To: <sip:0172xxxxxx@IP_HT503:5062>
Contact: <sip:0285xxxxxx@IP_Elastix>
Call-ID: 03cd9ff638061ea1534e2a5e2aca6af3@IP_HT503
CSeq: 102 INVITE
User-Agent: FPBX-2.6.0(1.4.36)
Max-Forwards: 70
Remote-Party-ID: "0285xxxxxx" <sip:0285xxxxxx@IP_HT503>;privacy=off;screen=no
Date: Thu, 19 Jun 2014 03:29:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 242

v=0
o=root 28905 28905 IN IP4 IP_Elastix
s=session
c=IN IP4 IP_Elastix
t=0 0
m=audio 10012 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called HT503/0172xxxxxx

<--- SIP read from IP_HT503:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP IP_Elastix:5060;branch=z9hG4bK7cd7cc1c;rport=5060
From: "0285xxxxxx" <sip:0285xxxxxx@IP_HT503>;tag=as1df6f147
To: <sip:0172xxxxxx@IP_HT503:5062>
Call-ID: 03cd9ff638061ea1534e2a5e2aca6af3@IP_HT503
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503  V1.1B 1.0.10.9  chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---

<--- SIP read from IP_HT503:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP IP_Elastix:5060;branch=z9hG4bK7cd7cc1c;rport=5060
From: "0285xxxxxx" <sip:0285xxxxxx@IP_HT503>;tag=as1df6f147
To: <sip:0172xxxxxx@IP_HT503:5062>;tag=350778873
Call-ID: 03cd9ff638061ea1534e2a5e2aca6af3@IP_HT503
CSeq: 102 INVITE
Contact: <sip:IP_HT503:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503  V1.1B 1.0.10.9  chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
    -- SIP/HT503-0000048e is ringing

<--- SIP read from IP_HT503:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP IP_Elastix:5060;branch=z9hG4bK7cd7cc1c;rport=5060
From: "0285xxxxxx" <sip:0285xxxxxx@IP_HT503>;tag=as1df6f147
To: <sip:0172xxxxxx@IP_HT503:5062>;tag=350778873
Call-ID: 03cd9ff638061ea1534e2a5e2aca6af3@IP_HT503
CSeq: 102 INVITE
Contact: <sip:IP_HT503:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503  V1.1B 1.0.10.9  chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length:   247

v=0
o=- 8002 8000 IN IP4 IP_HT503
s=SIP Call
c=IN IP4 IP_HT503
t=0 0
m=audio 10010 RTP/AVP 8 101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32-36,54
a=silenceSupp:off - - - -

<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port IP_HT503:10010
list_route: hop: <sip:IP_HT503:5062>
set_destination: Parsing <sip:IP_HT503:5062> for address/port to send to
set_destination: set destination to IP_HT503, port 5062
Transmitting (no NAT) to IP_HT503:5062:
ACK sip:IP_HT503:5062 SIP/2.0
Via: SIP/2.0/UDP IP_Elastix:5060;branch=z9hG4bK382385a7;rport
From: "0285xxxxxx" <sip:0285xxxxxx@IP_HT503>;tag=as1df6f147
To: <sip:0172xxxxxx@IP_HT503:5062>;tag=350778873
Contact: <sip:0285xxxxxx@IP_Elastix>
Call-ID: 03cd9ff638061ea1534e2a5e2aca6af3@IP_HT503
CSeq: 102 ACK
User-Agent: FPBX-2.6.0(1.4.36)
Max-Forwards: 70
Remote-Party-ID: "0285xxxxxx" <sip:0285xxxxxx@IP_HT503>;privacy=off;screen=no
Content-Length: 0
Ici je suis chez Free et je ne vois pas de d'adresse IP public Free.
Je ne sais pas comment tu as boutiqué ton install... Mais il y a quelque chose qui cloche.
J'ai mis le port FXO sur 5062

Mon trunk est comme ceci:
host=IP_HT503
fromdomain=IP_HT503
username=
secret=*****
type=peer
qualify=yes
progressinband=yes
disallow=all
allow=alaw
dtmfmode=RFC2833
insecure=port,invite
context=from-trunk
port=5062
canreinvite=no
call-limit=1
Click to expand...
 

olivier

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re,

un petit shema .

en gros ipbx ipphone et ht 5003 sur un hub , isole du reseau maison donc pas de pfsense ou autre firewall:
ip box 192.168.1.254/24
ip elastix 192.168.1.38/24 passerelle 192.168.1.61
ipphone 192.168.1.46/24 passerelle 192.168.1.61
ht 503 192.168.1.61/24passerelle 192.168.1.254
 

olivier

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petite chose

mon elastix est sur une alix board, j ai du mal a refaire la manip pour avoir une install clean donc je me sert d un backup easus qui avait les paramettre voip free.

donc sans passerelle psnt , juste voip ..

j essaye de faire une vm hyper-v mais ca marche pas, plantage..

je crois que je vais monter un pc pour ca...
a+
 
danardf

danardf

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Messages
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Hmmm possible.
J'ai un serveur sur un Netbook ça marche,
Enfin tout dépend du nombre de postes et de comm simultanées. :lol:
Mais pour un Athom 1,4Ghz 1Go de RAM, 15 postes ça dois passer à l'aise.
 

olivier

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Sep 13, 2013
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re,

il a t il un moyen de faire un reset to factory

j ai pas trouved option, mais peut etre une astuce

merci
 

olivier

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re,

ca marche, je te dirai demain ce que j ai fait car je suis creve et que je veux verifier certaine chose avant de m avancer et demander ton avis.
quel que debug sip et autre

merci
a+
 

olivier

Joined
Sep 13, 2013
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re,

je n ai pas modifie les fichier ht 503 dans un premier temps( ferais apres)

l erreur venait en fait (je pense ) de la config reseau mais pas sur elastix dans asterix sip sur freepbx.
je ne sais pas pourquoi encore, mais comme je te l avais dit, c'etait un backup d install du temps de mes test en voipfree.

ipbx etant derrier mon pfsense ca config etait en 192.168.2.2.
lorsque j ai isoler le resau voip comme conseille, j ai eu une adresse en 192.168.1.38.
ce qui ma mit la puce a l' oreil c'est ton commentaire sur l ip public.
comment ca se fait..
route sur elastix me donnait les bon paramettre.

j ai fait donc un tour sur asterix sip et la ben mon local network etait toujours en 192.168.2.0 et ip public celle de free.

j ai modifier lacal network en 192.168.1.0 et c'est passer.

mes debug sip etait correcte.

pourquoi l ip ne c'est pas mis a jour??

as tu une idée .

merci pour ton aide et ta patience.
a+
 
danardf

danardf

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Salut

Tu parles des paramètres NAT dans Asterisk Sip Settings de Freepbx?
Tant que tu ne cliqueras pas sur le bouton Auto-configure, l'adresse IP ne changera pas !!
 
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