Hi,<br><br>I am taking over an asterisk system from another person and having an issue where a sip trunk is restricting the outgoing codecs to just g.729<br><br>I am dialing in from a Cisco 7960. The Invite from the Cisco has the folowing M line:<br>
m=audio 17022 RTP/AVP 18 0 8 101.<br>So it is allowing g.729, ulaw and alaw.<br><br>Asterisk is tandeming the call out over a SIP trunk<br><br>sip.conf tandem trunk config:<br> [trunk-out]<br> host=192.168.1.6<br> type=friend <br>
disallow=all<br> allow=ulaw <br> allow=alaw<br> allow=gsm<br> allow=g729 <br> context=from-trunk<br> nat=no<br> qualify=100<br><br>But the outgoing Invite has the following m line:<br> m=audio 17064 RTP/AVP 18 101.<br>
<br>This system does realtime which I am not really familiar with but the only stuff that seems relivent is one table called sip_devices with 2 columns disallowd and allowed. I think this should only affect the phones though. For this extension the values are disallowed=all and allowed=g729;ulaw;alaw<br>
<br>I did try to search here and Google but I am not sure what to use for a search string.<br><br>I turned on debug to level 3 :<br><br> -- Executing [s@macro-dialout:36] Dial("SIP/1234-00000039", "SIP/trunkout/1xxxxxxxxx,60,L(180000:20000)") in new stack<br>
[Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:25057 sip_request_call: Asked to create a SIP channel with formats: 0x100 (g729)<br>[Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:7215 sip_alloc: Allocating new SIP dialog for <a href="http://4870fab953c16a611b9248584748fe59@127.0.0.1:0">4870fab953c16a611b9248584748fe59@127.0.0.1:0</a> - INVITE (No RTP)<br>
[Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:344 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x9ded230'<br>[Mar 19 18:22:56] DEBUG[17418]: res_rtp_asterisk.c:472 ast_rtp_new: Allocated port 19718 for RTP instance '0x9ded230'<br>
[Mar 19 18:22:56] DEBUG[17418]: rtp_engine.c:353 ast_rtp_instance_new: RTP instance '0x9ded230' is setup and ready to go<br>[Mar 19 18:22:56] DEBUG[17418]: res_rtp_asterisk.c:2370 ast_rtp_prop_set: Setup RTCP on RTP instance '0x9ded230'<br>
== Using SIP RTP TOS bits 184<br> == Using SIP RTP CoS mark 5<br>[Mar 19 18:22:56] DEBUG[17418]: chan_sip.c:4683 do_setnat: Setting NAT on RTP to Off<br>[Mar 19 18:22:57] DEBUG[17418]: acl.c:715 ast_ouraddrfor: For destination '192.168.1.6', our source address is '192.168.1.25'.<br>
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:3243 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address <a href="http://192.168.1.25:5060">192.168.1.25:5060</a><br>[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6557 sip_new: *** Our native formats are 0x100 (g729)<br>
[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6558 sip_new: *** Joint capabilities are 0x100 (g729)<br>[Mar 19 18:22:57] DEBUG[17418]: chan_sip.c:6559 sip_new: *** Our capabilities are 0x10e (gsm|ulaw|alaw|g729)<br><br>The receiving asterisk system only does ulaw or alaw.<br>
<br>I am sure it is a mis-configuration somewhere, just not finding it.<br><br>Where should I look to enable the other codecs?<br>What else would help in troubleshooting?<br><br>Thank you,<br>JH<br>