<div class="gmail_extra">Hi,</div><div class="gmail_extra"><br></div><div class="gmail_extra"><blockquote class="gmail_quote" style="margin-top:0px;margin-right:0px;margin-bottom:0px;margin-left:0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">
<span style>No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'</span> </blockquote><div> </div><div>This line is telling you everything. The peer you've declared isn't being matched for the incoming call and hence it tries to look in "default" context (I assume allowguest=yes in your sip.conf)</div>
<div><br></div><div>Make sure that your peer is matched, since you've qualify=yes defined execute the command "sip show peer Trunk-Telco" in asterisl CLI and see the status of the peer.</div><div><br></div><div>
What I'm guessing is that the telco has multiple IPs to send you calls and the incoming call isn't coming from the IP you've declared in your sip telco-trunk section. I don't think we can set a subnet in host=87.XX.XX.XX/28 parameter.!!</div>
<div><br></div><div>Regards,</div><div>Sammy.</div><br><div class="gmail_quote">On Sat, Apr 21, 2012 at 11:47 AM, Michel Verbraak <span dir="ltr"><<a href="mailto:michel@verbraak.org" target="_blank">michel@verbraak.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="HOEnZb"><div class="h5">On 21-04-12 08:19, Olivier CALVANO wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi<br>
<br>
I have a small problems with incoming call.<br>
<br>
I have a peer actually configured for outcall :<br>
<br>
<br>
sip.conf:<br>
<br>
[Trunk-Telco]<br>
type=peer<br>
host=<a href="http://domaineofmysupplier.net" target="_blank">domaineofmysupplier.net</a><br>
outboundproxy=<a href="http://domaineofmysupplier.net" target="_blank">domaineofmysuppl<u></u>ier.net</a><br>
session-timers=originate<br>
session-expires=7200<br>
qualify=yes<br>
dtmf=rfc2833<br>
nat=no<br>
canreinvite=no<br>
canredirect=yes<br>
dtmfmode=rfc2833<br>
disallow=all<br>
allow=alaw<br>
insecure=port,invite<br>
context=incoming<br>
<br>
This SIP account work for outgoing call. when i want receive call from<br>
this sipplier, i have a "extension not found".<br>
<br>
In extensions.conf for incoming:<br>
<br>
[incoming]<br>
exten => _X.,1,Dial(IAX2/VoIP/${EXTEN},<u></u>180,rt)<br>
<br>
in dialplan show incoming, no problems i see the dialplan.<br>
<br>
when i call, i have:<br>
<br>
<--- SIP read from UDP://84.xx.xx.72:5060 ---><br>
INVITE sip:331NUMNOFOUND@78.<u></u>IPOFMYSERVER:5060 SIP/2.0<br>
Record-Route:<sip:84.xx.xx.72;<u></u>r2=on;lr;f=4><br>
Record-Route:<sip:172.16.21.<u></u>172;r2=on;lr;f=4><br>
Record-Route:<sip:172.16.21.<u></u>67;lr;f=8><br>
Record-Route:<sip:172.16.20.<u></u>119;lr;did=247.29f60367><br>
Via: SIP/2.0/UDP 84.xx.xx.72;branch=<u></u>z9hG4bK10e4.e7f23f11.0<br>
Via: SIP/2.0/UDP 172.16.21.67;branch=<u></u>z9hG4bK10e4.bbf4c444.0<br>
Via: SIP/2.0/UDP 172.16.20.119;branch=<u></u>z9hG4bK10e4.9fe53c91.0<br>
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=<u></u>z9hG4bK00151747E2606DB6CA39464<u></u>AF542<br>
From: "+331MYCLID"<br>
<sip:+331MYCLID;tgrp=<a href="mailto:RT43@172.16.21.11" target="_blank">RT43@172.<u></u>16.21.11</a>>;tag=<u></u>2RUVP51HBW30000E1D00001u0K4NFQ<u></u>C0QNAN31<br>
To:<<a href="mailto:sip%3A%2B331NUMNOFOUND@172.16.20.119" target="_blank">sip:+331NUMNOFOUND@172.16.<u></u>20.119</a>><br>
Call-ID: <a href="mailto:60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1" target="_blank">60471500e217-4f924d2c-<u></u>477df10c-66ea6f8-140732f@127.<u></u>0.0.1</a><br>
CSeq: 20114 INVITE<br>
Contact:<<a href="http://sip:+331MYCLID@172.16.21.11:5060" target="_blank">sip:+331MYCLID@172.<u></u>16.21.11:5060</a>><br>
Allow-Events: refer<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,<br>
SUBSCRIBE, UPDATE<br>
Content-Type: application/sdp<br>
Max-Forwards: 67<br>
P-Asserted-Identity:<<a href="mailto:sip%3A%2B331MYCLID@domaineofmysupplier.net" target="_blank">sip:+<u></u>331MYCLID@domaineofmysupplier.<u></u>net</a>><br>
Supported: timer, replaces<br>
Content-Length: 369<br>
Min-SE: 90<br>
Session-Expires: 300<br>
P-Charging-Vector: icid-value="<a href="mailto:4f924d2c1e20abe1d@172.16.20.119" target="_blank">4f924d2c1e20abe1d@<u></u>172.16.20.119</a>"<br>
X-PSN-Trunk: ME<br>
<br>
v=0<br>
o=- 18406958643964291255 1 IN IP4 172.16.21.11<br>
s=session<br>
c=IN IP4 84.xx.xx.34<br>
t=0 0<br>
m=audio 64296 RTP/AVP 8 18 4 0 105 101<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:18 G729/8000<br>
a=rtpmap:4 G723/8000<br>
a=fmtp:4 annexa=no<br>
a=fmtp:4 bitrate=6.3<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:105 X-CCD/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=ptime:20<br>
a=sendrecv<br>
a=nortpproxy:yes<br>
<br>
<-------------><br>
--- (25 headers 17 lines) ---<br>
== Using SIP RTP CoS mark 5<br>
Sending to 84.xx.xx.72 : 5060 (no NAT)<br>
Using INVITE request as basis request -<br>
<a href="mailto:60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1" target="_blank">60471500e217-4f924d2c-<u></u>477df10c-66ea6f8-140732f@127.<u></u>0.0.1</a><br>
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'<br>
Found RTP audio format 8<br>
Found RTP audio format 18<br>
Found RTP audio format 4<br>
Found RTP audio format 0<br>
Found RTP audio format 105<br>
Found RTP audio format 101<br>
Peer audio RTP is at port 84.xx.xx.34:64296<br>
Found audio description format PCMA for ID 8<br>
Found audio description format G729 for ID 18<br>
Found audio description format G723 for ID 4<br>
Found audio description format PCMU for ID 0<br>
Found unknown media description format X-CCD for ID 105<br>
Found audio description format telephone-event for ID 101<br>
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d<br>
(g723|ulaw|alaw|g729)/video=<u></u>0x0 (nothing)/text=0x0 (nothing), combined<br>
- 0xc (ulaw|alaw)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1<br>
(telephone-event), combined - 0x1 (telephone-event)<br>
Peer audio RTP is at port 84.xx.xx.34:64296<br>
Looking for 331NUMNOFOUND in default (domain 78.IPOFMYSERVER)<br>
</blockquote></div></div>
It is looking for the 331NUMNOFOUND in context named "default".<br>
Do you have this context? Does the extension exists in the context?<br>
<br>
Do you have a register line in your sip.conf for this external provider? In the register line you can specify the extensions/device to use in the sip.conf so it knows the right context to start in extensions.conf instead of the default context.<br>
<br>
For example: register => <a href="http://username:password@sip.voipbuster.com/Trunk-Telco" target="_blank">username:password@sip.<u></u>voipbuster.com/Trunk-Telco</a><div class="im"><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
<--- Reliably Transmitting (no NAT) to 84.xx.xx.72:5060 ---><br>
SIP/2.0 404 Not Found<br>
Via: SIP/2.0/UDP 84.xx.xx.72;branch=<u></u>z9hG4bK10e4.e7f23f11.0;<u></u>received=84.xx.xx.72<br>
</blockquote></div>
<snip><div class="im"><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<------------><br>
[Apr 21 08:01:16] NOTICE[11906]: chan_sip.c:18527<br>
handle_request_invite: Call from '' to extension '331NUMNOFOUND'<br>
rejected because extension not found.<br>
<br>
<br>
<br>
<br>
<br>
</blockquote></div>
Regards,<br>
Michel.<div class="HOEnZb"><div class="h5"><br>
<br>
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</div></div></blockquote></div><br></div>