No luck so far, should I consider it a bug in Asterisk 11 as I have tried different version of Asterisk 11 as well. Carrier sends BYE with service not implemented where as asterisk advertise udptl in SDP for Answer. I do not want it to be advertised by Asterisk 11 in Answer() as I am not using it(udptl, fax etc) in any case.<div>
<br><br><div class="gmail_quote">On Thu, Jan 17, 2013 at 11:15 AM, Salman Zafar <span dir="ltr"><<a href="mailto:msalman212@gmail.com" target="_blank">msalman212@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Thanks <span name="Matthew Jordan" style="vertical-align:top;color:rgb(34,34,34);font-size:13px;white-space:nowrap;font-family:arial,sans-serif;display:inline">Jordan, for having a look at this matter.</span><div>
<span name="Matthew Jordan" style="vertical-align:top;color:rgb(34,34,34);font-size:13px;white-space:nowrap;font-family:arial,sans-serif;display:inline"><br></span></div><div><span name="Matthew Jordan" style="vertical-align:top;color:rgb(34,34,34);font-size:13px;white-space:nowrap;font-family:arial,sans-serif;display:inline">Yes, that is what Asterisk 11 is sending. Here are complete sip debugs from Asterisk attached. Please refer to IP mapping from OP to have a better understanding.</span></div>
<div><span name="Matthew Jordan" style="vertical-align:top;color:rgb(34,34,34);font-size:13px;white-space:nowrap;font-family:arial,sans-serif;display:inline"><br></span></div><div>
<span name="Matthew Jordan" style="vertical-align:top;color:rgb(34,34,34);font-size:13px;white-space:nowrap;font-family:arial,sans-serif;display:inline">Is there any way of getting it off from SIP parser on compile time as I am not using this feature and do not intend to use in future.</span></div>
<div><font color="#222222" face="arial, sans-serif"><span style="white-space:nowrap"><br></span></font></div><div><div><div class="h5"><font color="#222222" face="arial, sans-serif"><span style="white-space:nowrap"><br></span></font><br>
<div class="gmail_quote">
On Wed, Jan 16, 2013 at 7:01 PM, Matthew Jordan <span dir="ltr"><<a href="mailto:mjordan@digium.com" target="_blank">mjordan@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>On 01/16/2013 07:28 AM, Salman Zafar wrote:<br>
> Hello All,<br>
> I am having a bit peculiar problem with Asterisk 11 for a<br>
> carrier. This carrier shares quite some information in SDP header, which<br>
> should not be the problem, however what happen is as follow:<br>
><br>
><br>
</div>> Carrier----> (INVITE) -> *SIP Proxy -> Asterisk 11 -> Answer()* -> right<br>
<div>> after answering call drops... Carrier send a BYE with (cause 79: service<br>
> or option not implemented).<br>
><br>
</div>> *NOTE: Please refer to complete SIP traces attached. *<br>
> *<br>
> *<br>
> *Also Note:*<br>
> _Carrier_: 62.61.147.214<br>
> _Proxy_: 77.X.X.X:5060<br>
> _Asterisk11_: 77.X.X.X:5080<br>
><br>
> *_Here is Invite SDP from Carrier -> Proxy -> Asterisk 11_*<br>
<div>><br>
> INVITE sip:69609000@77.X.X.X SIP/2.0<br>
> v=0<br>
> o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214<br>
> s=Phone-Call<br>
> c=IN IP4 77.X.X.X<br>
> t=0 0<br>
> m=audio 53372 RTP/AVP 8 118 18<br>
> a=rtpmap:8 PCMA/8000<br>
> a=rtpmap:118 PCMA/8000<br>
> a=gpmd:118 vbd=yes<br>
> a=rtpmap:18 G729/8000<br>
> a=fmtp:18 annexb=no<br>
> a=ptime:20<br>
> a=sendrecv<br>
> a=rtcp:53373 IN IP4 77.X.X.X<br>
> m=image 56854 udptl t38<br>
> a=T38FaxVersion:0<br>
> a=T38MaxBitRate:14400<br>
> a=T38FaxMaxBuffer:1024<br>
> a=T38FaxMaxDatagram:122<br>
> a=T38FaxRateManagement:transferredTCF<br>
> a=T38FaxUdpEC:t38UDPRedundancy<br>
><br>
</div>> /*_SDP:After Answered by Asterisk 11_*/<br>
<div>> v=0<br>
> o=root 164966782 164966782 IN IP4 77.X.X.X<br>
> s=Asterisk v11.0.1<br>
> c=IN IP4 77.X.X.X<br>
> t=0 0<br>
> m=audio 12636 RTP/AVP 18 8<br>
> a=rtpmap:18 G729/8000<br>
> a=fmtp:18 annexb=no<br>
> a=rtpmap:8 PCMA/8000<br>
> a=ptime:20<br>
> a=sendrecv<br>
</div>> *_m=image 0 udptl t38_*<br>
<br>
<br>
The appropriate way for Asterisk to indicate that it does not support a<br>
media stream is to set the port number to 0. We have to inform the<br>
offerer that we don't support the media stream; removing it from the SDP<br>
completely is not allowed.<br>
<br>
Per RFC 3264, section 6:<br>
<br>
" An offered stream MAY be rejected in the answer, for any reason. If<br>
a stream is rejected, the offerer and answerer MUST NOT generate<br>
media (or RTCP packets) for that stream. To reject an offered<br>
stream, the port number in the corresponding stream in the answer<br>
MUST be set to zero. "<br>
<br>
> I have tired by disabling/unloading fax modules as *I am not using* them<br>
<div>> but no results. Secondly, also tried tweaking of udptl ever-odd nothing<br>
> worked.<br>
<br>
</div>You've configured your system to not support fax correctly. Asterisk is<br>
rejecting the offered image stream accordingly.<br>
<div><br>
> The same carrier works for Asterisk 1.6.X and the only difference I have<br>
> notice so far is the above underlined line in Answered SDP -> m=image 0<br>
> udptl t38. I think if I some how do not advertise udptl here i would be<br>
> able to avoid this scenario. I have tried multiple ways to strip off SDP<br>
> from incoming INVITE at SIP Proxy level but it is not SDP wise enough.<br>
><br>
<br>
</div>I'm not sure what 1.6.x is sending. It's possible that it just<br>
completely removed the stream from the SDP answer, which is wrong.<br>
<br>
Section 6 again:<br>
<br>
"For each "m=" line in the offer, there MUST be a corresponding "m="<br>
line in the answer."<br>
<br>
> *Note:*<br>
<div>><br>
> In Asterisk 1.6 => WARNING[32671]: chan_sip.c:8833 process_sdp:<br>
> Unsupported SDP media type in offer: image 59978 udptl t38<br>
> In Asterisk 11 => WARNING[18748][C-0000002f]: chan_sip.c:10277<br>
> process_sdp: Failed to initialize UDPTL, declining image stream<br>
><br>
><br>
<br>
</div>An initial glance at this makes me think your carrier is doing something<br>
wrong. Just to check, however, is the SDP answer you pasted the entire<br>
SDP that Asterisk 11 responds with? Specifically, are there no format<br>
attributes for the image stream in the SDP that Asterisk responds with?<br>
<br>
--<br>
Matthew Jordan<br>
Digium, Inc. | Engineering Manager<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br>
<br>
<br>
<br>
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</blockquote></div><br><br clear="all"><div><br></div></div></div><span class="HOEnZb"><font color="#888888">-- <br><font face="'times new roman', serif">Regards</font><div><font face="'times new roman', serif"><br>
</font><div><pre cols="72"><font face="'times new roman', serif">**************************
Muhammad Salman
***************************</font>
</pre></div></div>
</font></span></div>
</blockquote></div><br><br clear="all"><div><br></div>-- <br><font face="'times new roman', serif">Regards</font><div><font face="'times new roman', serif"><br></font><div><pre cols="72"><font face="'times new roman', serif">**************************
Muhammad Salman
***************************</font>
</pre></div></div>
</div>