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<p>I am having a problem with one of my callers who is using either
g729 or alaw. I can do alaw but not g729 so asterisk should
negotiate alaw right? In fact from the sip debug it looks like it
does, but then I get the dreaded "channel.c:5630 set_format:
Unable to find a codec translation path: (g729) -> (alaw)" and
the call hangs up. Why?</p>
<p>Last minute thought: Is it possible that the caller is sending
g729 in RTP even though the SIP negotiation clearly chooses alaw?
Maybe I need some RTP debugging.<br>
</p>
<p>Asterisk 13.14.1 on Debian, using chan_sip.</p>
<p>Here's the trace:</p>
<pre><--- SIP read from UDP:SUPPLIER:5060 --->
INVITE <a class="moz-txt-link-freetext" href="sip:LOCAL@ASTERISK:5060" moz-do-not-send="true">sip:LOCAL@ASTERISK:5060</a> SIP/2.0
Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9
From: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER" moz-do-not-send="true"><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK" moz-do-not-send="true"><sip:LOCAL@ASTERISK></a>
Call-ID: 205665777_90679951@SUPPLIER
CSeq: 539098 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER:5060" moz-do-not-send="true"><sip:REMOTE@SUPPLIER:5060></a>
P-Asserted-Identity: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@REMOTE-SUPPLIER;user=phone" moz-do-not-send="true"><sip:REMOTE@REMOTE-SUPPLIER;user=phone></a>
Supported: timer,100rel,precondition
Session-Expires: 1800
Min-SE: 90
Content-Length: 282
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 176880 320591 IN IP4 SUPPLIER
s=SIP Media Capabilities
c=IN IP4 213.41.124.6
t=0 0
m=audio 8526 RTP/AVP 18 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
<------------->
--- (17 headers 13 lines) ---
Sending to SUPPLIER:5060 (no NAT)
Sending to SUPPLIER:5060 (no NAT)
Using INVITE request as basis request - 205665777_90679951@SUPPLIER
Found peer 'supplier' for 'REMOTE' from SUPPLIER:5060
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|gsm), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 213.41.124.6:8526
Looking for LOCAL in supplier-in (domain ASTERISK)
sip_route_dump: route/path hop: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER:5060" moz-do-not-send="true"><sip:REMOTE@SUPPLIER:5060></a>
</pre>
<blockquote>
<p>So, all looking good here, we've worked out that the combined
capabilities are (alaw)<br>
</p>
</blockquote>
<pre><--- Transmitting (no NAT) to SUPPLIER:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER
From: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER" moz-do-not-send="true"><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK" moz-do-not-send="true"><sip:LOCAL@ASTERISK></a>
Call-ID: 205665777_90679951@SUPPLIER
CSeq: 539098 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK:5060" moz-do-not-send="true"><sip:LOCAL@ASTERISK:5060></a>
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to SUPPLIER:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER
From: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER" moz-do-not-send="true"><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK" moz-do-not-send="true"><sip:LOCAL@ASTERISK></a>;tag=as4502927f
Call-ID: 205665777_90679951@SUPPLIER
CSeq: 539098 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK:5060" moz-do-not-send="true"><sip:LOCAL@ASTERISK:5060></a>
Content-Length: 0
<------------>
Audio is at 13948
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to SUPPLIER:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER
From: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER" moz-do-not-send="true"><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK" moz-do-not-send="true"><sip:LOCAL@ASTERISK></a>;tag=as4502927f
Call-ID: 205665777_90679951@SUPPLIER
CSeq: 539098 INVITE
Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK:5060" moz-do-not-send="true"><sip:LOCAL@ASTERISK:5060></a>
Content-Type: application/sdp
Require: timer
Content-Length: 264
v=0
o=root 227409966 227409966 IN IP4 ASTERISK
s=Asterisk PBX 13.14.1~dfsg-2+deb9u4
c=IN IP4 ASTERISK
t=0 0
m=audio 13948 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
</pre>
<blockquote>
<p>And that's good to, we've sent the OK for the INVITE saying
that we want alaw.</p>
<p><br>
</p>
</blockquote>
<pre><--- SIP read from UDP:SUPPLIER:5060 --->
ACK <a class="moz-txt-link-freetext" href="sip:LOCAL@ASTERISK:5060" moz-do-not-send="true">sip:LOCAL@ASTERISK:5060</a> SIP/2.0
Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5bc037285f864da9
From: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER" moz-do-not-send="true"><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK" moz-do-not-send="true"><sip:LOCAL@ASTERISK></a>;tag=as4502927f
Call-ID: 205665777_90679951@SUPPLIER
CSeq: 539098 ACK
Max-Forwards: 70
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
[May 13 13:46:58] WARNING[7245][C-000031da]: channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)
</pre>
<blockquote>
<p>What's this nonsense! Why is set_format trying to use g729!<br>
</p>
</blockquote>
<pre>Scheduling destruction of SIP dialog '205665777_90679951@SUPPLIER' in 32000 ms (Method: ACK)
set_destination: Parsing <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER:5060" moz-do-not-send="true"><sip:REMOTE@SUPPLIER:5060></a> for address/port to send to
set_destination: set destination to SUPPLIER:5060
Reliably Transmitting (no NAT) to SUPPLIER:5060:
BYE <a class="moz-txt-link-freetext" href="sip:REMOTE@SUPPLIER:5060" moz-do-not-send="true">sip:REMOTE@SUPPLIER:5060</a> SIP/2.0
Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d
Max-Forwards: 70
From: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK" moz-do-not-send="true"><sip:LOCAL@ASTERISK></a>;tag=as4502927f
To: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER" moz-do-not-send="true"><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
Call-ID: 205665777_90679951@SUPPLIER
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:SUPPLIER:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d
From: <a class="moz-txt-link-rfc2396E" href="sip:LOCAL@ASTERISK" moz-do-not-send="true"><sip:LOCAL@ASTERISK></a>;tag=as4502927f
To: <a class="moz-txt-link-rfc2396E" href="sip:REMOTE@SUPPLIER" moz-do-not-send="true"><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
Call-ID: 205665777_90679951@SUPPLIER
CSeq: 102 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '205665777_90679951@SUPPLIER' Method: ACK
</pre>
<p><br>
</p>
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