siptrunk1 <-> asterisk <-> siptrunk2

Asterisk Asterisk Support
1

Hello,

I’m trying to do a simple setup in which asterisk will forward calls between multiple trunk.

At first, I’m trying to make it forward calls between 2 sip trunks but without success.
Everything is on the same local network, no nat, asterisk 11.0.1.

Here is my sip.conf :

[general]                                                                      
udpbindaddr=0.0.0.0:6060                                            
tcpenable=yes                     ; one of my sip trunk use tcp, no choice here                       
tcpbindaddr=127.0.0.1:6061                             
disallow=all                                                                   
allow=speex,gsm,ulaw,alaw                                                           
sipdebug=yes                                                                   


[siproxd]                                       ; that one will be always here, it's a patched siproxd
type=friend                                                                    
host=localhost                                                                 
outboundproxy=tcp://127.0.0.1:5060                                             
context=from-siproxd                                                           
transport=tcp                                                                  
insecure=port,invite                                                           

[ipbx]   ; the other trunk, a kamilio 3.3.0 with dispatcher module for my test
type=friend                                                                    
host=192.168.10.44
outboundproxy=udp://192.168.10.44:6060
context=from-ipbx                                                              
insecure=port,invite

and my extension.conf

[from-siproxd]
exten => _X.,1,Dial(SIP/ipbx/${EXTEN},30,r)

[from-ipbx]
exten => _X.,1,Dial(SIP/siproxd/${EXTEN},30,r)

When I’m placing call from my “ipbx”, I got the following log output :

...received SIP invite...
[Feb 12 16:43:44] VERBOSE[7600][C-00000000] chan_sip.c: Using INVITE request as basis request - 223887008
[Feb 12 16:43:44] VERBOSE[7600][C-00000000] chan_sip.c: Found peer 'ipbx' for '33600000002' from 192.168.10.44:6060
...sdp related stuff...
[Feb 12 16:43:44] VERBOSE[7600][C-00000000] chan_sip.c: Peer audio RTP is at port 192.168.10.106:7078
[Feb 12 16:43:44] VERBOSE[7600][C-00000000] chan_sip.c: Looking for 3000 in from-ipbx (domain trantor)
... sip 404 not found is sent to ipbx...
[Feb 12 16:43:44] NOTICE[7600][C-00000000] chan_sip.c: Call from 'ipbx' (192.168.10.44:6060) to extension '3000' rejected because extension not found in context 'from-ipbx'.
[Feb 12 16:43:44] VERBOSE[7600][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '223887008' in 32000 ms (Method: INVITE)

and if I’m trying to place a call from “siproxd” to “ipbx” I got the same error :

...received SIP invite...
[Feb 12 16:52:59] VERBOSE[7604][C-00000001] chan_sip.c: Found peer 'siproxd' for '3000' from 127.0.0.1:34254
...sdp related stuff...
[Feb 12 16:54:01] VERBOSE[7600][C-00000002] chan_sip.c: Peer audio RTP is at port 192.168.10.106:7078
[Feb 12 16:52:59] VERBOSE[7604][C-00000001] chan_sip.c: Looking for 33600000002 in from-siproxd (domain 192.168.10.80)
... sip 404 not found is sent to siproxd...
[Feb 12 16:52:59] NOTICE[7604][C-00000001] chan_sip.c: Call from 'siproxd' (127.0.0.1:34254) to extension '33600000002' rejected because extension not found in context 'from-siproxd'.
[Feb 12 16:52:59] VERBOSE[7604][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '749175007' in 32000 ms (Method: INVITE)

Anyone have any idea of what I’m doing wrong or have any clue on how to debug it ?

thanks very much for any help

2

Do dialplan show to verify extensions.conf has read correctly.

(You may want to review your sip.conf type and insecure parameters.)