<div dir="ltr"><div>Argh. That was for chan_pjsip and you are using chan_sip. Be aware that chan_sip is effectively dead.<br></div><div><br></div><div>Richard<br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, May 14, 2020 at 9:50 AM Richard Mudgett <<a href="mailto:rmudgett@digium.com">rmudgett@digium.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div>The other end is sending g729 even though it was not negotiated. The other end should not do this and it usually seems that the other ends that do send g729.<br></div><div>This was recently fixed. See <a href="https://issues.asterisk.org/jira/browse/ASTERISK-28139" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-28139</a></div><div><br></div><div>Richard<br></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, May 14, 2020 at 1:11 AM John Hughes <<a href="mailto:john@calva.com" target="_blank">john@calva.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<div>
<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>sip:LOCAL@ASTERISK:5060</a> SIP/2.0
Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9
From: <a><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a><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><sip:REMOTE@SUPPLIER:5060></a>
P-Asserted-Identity: <a><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 <a href="http://213.41.124.6:8526" target="_blank">213.41.124.6:8526</a>
Looking for LOCAL in supplier-in (domain ASTERISK)
sip_route_dump: route/path hop: <a><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><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a><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><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><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a><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><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><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a><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><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>sip:LOCAL@ASTERISK:5060</a> SIP/2.0
Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5bc037285f864da9
From: <a><sip:REMOTE@SUPPLIER></a>;tag=gK02498cb1
To: <a><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><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>sip:REMOTE@SUPPLIER:5060</a> SIP/2.0
Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d
Max-Forwards: 70
From: <a><sip:LOCAL@ASTERISK></a>;tag=as4502927f
To: <a><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><sip:LOCAL@ASTERISK></a>;tag=as4502927f
To: <a><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>
</div>
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