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you must have a way to debug gateway side, have a look there, sip trace
doesn't give me any idea.<br>
<br>
Olivier<br>
<br>
flavio a écrit :
<blockquote
cite="mid85ec7b250705310906k49d8f0e7t20a3c9760755c98e@mail.gmail.com"
type="cite">
<pre wrap="">---------- Forwarded message ----------
From: flavio <a class="moz-txt-link-rfc2396E" href="mailto:flavio.patria@gmail.com"><flavio.patria@gmail.com></a>
Date: 31-mag-2007 18.04
Subject: Re: [Serusers] SIP 479 Regretfully
To: <a class="moz-txt-link-abbreviated" href="mailto:olivier.taylor@hh174.be">olivier.taylor@hh174.be</a>
But If I try with my BT102 GrandStream (configured as Polycom on my
SER) I'm able to start a call to PSTN Number.
How is it possible? Have you any suggestions about?
Thanks,
ps I've also asterisk running on the same machine listening on port 5062.
U 2007/05/31 18:00:52.457140 10.28.19.124:5060 -> 10.28.19.202:5060
INVITE <a class="moz-txt-link-freetext" href="sip:0672020949@10.28.19.202">sip:0672020949@10.28.19.202</a> SIP/2.0.
Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bK5654e41ba6af4166.
From: <a class="moz-txt-link-rfc2396E" href="sip:0660522016@10.28.19.202"><sip:0660522016@10.28.19.202></a>;tag=44dcaa39db672de9.
To: <a class="moz-txt-link-rfc2396E" href="sip:0672020949@10.28.19.202"><sip:0672020949@10.28.19.202></a>.
Contact: <a class="moz-txt-link-rfc2396E" href="sip:0660522016@10.28.19.124"><sip:0660522016@10.28.19.124></a>.
Supported: replaces.
Proxy-Authorization: Digest username="0660522016",
realm="10.28.19.202", algorithm=MD5,
uri=<a class="moz-txt-link-rfc2396E" href="sip:0672020949@10.28.19.202">"sip:0672020949@10.28.19.202"</a>,
nonce="465f0e8002ccdf57a20d382a49036bd0e5c691d0",
response="554b818d1ecfc8455f1bc2e774881508".
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:19d87a9eb5a4d432@10.28.19.124">19d87a9eb5a4d432@10.28.19.124</a>.
CSeq: 17072 INVITE.
User-Agent: Grandstream BT110 1.0.8.12.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 211.
.
v=0.
o=0660522016 8000 8001 IN IP4 10.28.19.124.
s=SIP Call.
c=IN IP4 10.28.19.124.
t=0 0.
m=audio 5004 RTP/AVP 18 8 0.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
2007/5/31, olivier.taylor <a class="moz-txt-link-rfc2396E" href="mailto:olivier.taylor@gmail.com"><olivier.taylor@gmail.com></a>:
</pre>
<blockquote type="cite">
<pre wrap="">hi,
Regretfully, we were not able to process the URI
probably a malformed URI, don't you have to remove the leading 0 and add the international code?
hope it helps,
Olivier
flavio a écrit :
</pre>
<blockquote type="cite">
<pre wrap="">Hi to all.
I've configured my polycom ip500 IPphone to use it with ser.
If I try a call to users registred to ser all works fine.
If I try to call PSTN Number through my gateway I've the follow sip message:
INVITE <a class="moz-txt-link-freetext" href="sip:0672028405@10.28.19.202:5060;user=phone">sip:0672028405@10.28.19.202:5060;user=phone</a> SIP/2.0.
Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD.
From: "0660522015" <a class="moz-txt-link-rfc2396E" href="sip:0660522015@10.28.19.202"><sip:0660522015@10.28.19.202></a>;tag=8DDF0DB-E8BCEB84.
To: <a class="moz-txt-link-rfc2396E" href="sip:0672028405@10.28.19.202;user=phone"><sip:0672028405@10.28.19.202;user=phone></a>.
CSeq: 2 INVITE.
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:264aa927-d2a3a7c9-64f3fe3a@10.28.19.143">264aa927-d2a3a7c9-64f3fe3a@10.28.19.143</a>.
Contact: <a class="moz-txt-link-rfc2396E" href="sip:0660522015@10.28.19.143"><sip:0660522015@10.28.19.143></a>.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER.
User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1.
Supported: 100rel,replace.
Allow-Events: talk,hold,conference.
Proxy-Authorization: Digest username="0660522015",
realm="10.28.19.202",
nonce="465ef44bbb09c0155dc8555314519a20cac896f8",
uri=<a class="moz-txt-link-rfc2396E" href="sip:0672028405@10.28.19.202:5060;user=phone">"sip:0672028405@10.28.19.202:5060;user=phone"</a>,
response="e6eed258e095ee6be5be6c92210f9d99", algorithm=MD5.
Max-Forwards: 70.
Content-Type: application/sdp.
Content-Length: 237.
.
v=0.
o=- 1180620549 1180620549 IN IP4 10.28.19.143.
s=Polycom IP Phone.
c=IN IP4 10.28.19.143.
t=0 0.
m=audio 2234 RTP/AVP 18 8 0 101.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
#
U 2007/05/31 16:09:03.527834 10.28.19.202:5060 -> 10.28.19.143:5060
SIP/2.0 479 Regretfully, we were not able to process the URI (479/SL).
Via: SIP/2.0/UDP 10.28.19.143;branch=z9hG4bKbb2623fe3F3B54DD.
From: "0660522015" <a class="moz-txt-link-rfc2396E" href="sip:0660522015@10.28.19.202"><sip:0660522015@10.28.19.202></a>;tag=8DDF0DB-E8BCEB84.
To: <a class="moz-txt-link-rfc2396E" href="sip:0672028405@10.28.19.202;user=phone"><sip:0672028405@10.28.19.202;user=phone></a>;tag=979d95a734c13f6db8b9e3a72b9f44a0.14e7.
CSeq: 2 INVITE.
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:264aa927-d2a3a7c9-64f3fe3a@10.28.19.143">264aa927-d2a3a7c9-64f3fe3a@10.28.19.143</a>.
Server: Sip EXpress router (0.9.6 (i386/linux)).
Content-Length: 0.
Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=8314
req_src_ip=10.28.19.143 req_src_port=5060
in_uri=<a class="moz-txt-link-freetext" href="sip:0672028405@10.28.19.202:5060;user=phone">sip:0672028405@10.28.19.202:5060;user=phone</a>
out_uri=<a class="moz-txt-link-freetext" href="sip:0672028405@10.28.52.105:5060:5060;user=phone">sip:0672028405@10.28.52.105:5060:5060;user=phone</a> via_cnt==1".
Have you any suggestion about?
I use my Polycom with Asterisk and BroadSoft without any problem.
Thanks for your support and great patience :D
Bye,
F.
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</pre>
</blockquote>
</blockquote>
<pre wrap=""><!---->
--
********************************
* (o< ing. Patria Flavio
* //\ phone 0823451358
* V_/_ mobile 3407873357
*
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</pre>
</blockquote>
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