[asterisk-users] No extension found ?

Olivier CALVANO o.calvano at gmail.com
Sat Apr 21 01:19:09 CDT 2012


Hi

I have a small problems with incoming call.

I have a peer actually configured for outcall :


sip.conf:

[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=port,invite
context=incoming

This SIP account work for outgoing call. when i want receive call from
this sipplier, i have a "extension not found".

In extensions.conf for incoming:

[incoming]
        exten => _X.,1,Dial(IAX2/VoIP/${EXTEN},180,rt)

in dialplan show incoming, no problems i see the dialplan.

when i call, i have:

<--- SIP read from UDP://84.xx.xx.72:5060 --->
INVITE sip:331NUMNOFOUND at 78.IPOFMYSERVER:5060 SIP/2.0
Record-Route: <sip:84.xx.xx.72;r2=on;lr;f=4>
Record-Route: <sip:172.16.21.172;r2=on;lr;f=4>
Record-Route: <sip:172.16.21.67;lr;f=8>
Record-Route: <sip:172.16.20.119;lr;did=247.29f60367>
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: "+331MYCLID"
<sip:+331MYCLID;tgrp=RT43 at 172.16.21.11>;tag=2RUVP51HBW30000E1D00001u0K4NFQC0QNAN31
To: <sip:+331NUMNOFOUND at 172.16.20.119>
Call-ID: 60471500e217-4f924d2c-477df10c-66ea6f8-140732f at 127.0.0.1
CSeq: 20114 INVITE
Contact: <sip:+331MYCLID at 172.16.21.11:5060>
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Max-Forwards: 67
P-Asserted-Identity: <sip:+331MYCLID at domaineofmysupplier.net>
Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value="4f924d2c1e20abe1d at 172.16.20.119"
X-PSN-Trunk: ME

v=0
o=- 18406958643964291255 1 IN IP4 172.16.21.11
s=session
c=IN IP4 84.xx.xx.34
t=0 0
m=audio 64296 RTP/AVP 8 18 4 0 105 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=fmtp:4 bitrate=6.3
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<------------->
--- (25 headers 17 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 84.xx.xx.72 : 5060 (no NAT)
Using INVITE request as basis request -
60471500e217-4f924d2c-477df10c-66ea6f8-140732f at 127.0.0.1
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 101
Peer audio RTP is at port 84.xx.xx.34:64296
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Found unknown media description format X-CCD for ID 105
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined
- 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 84.xx.xx.34:64296
Looking for 331NUMNOFOUND in default (domain 78.IPOFMYSERVER)

<--- Reliably Transmitting (no NAT) to 84.xx.xx.72:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0;received=84.xx.xx.72
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: "+331MYCLID"
<sip:+331MYCLID;tgrp=RT43 at 172.16.21.11>;tag=2RUVP51HBW30000E1D00001u0K4NFQC0QNAN31
To: <sip:+331NUMNOFOUND at 172.16.20.119>;tag=as53fc96aa
Call-ID: 60471500e217-4f924d2c-477df10c-66ea6f8-140732f at 127.0.0.1
CSeq: 20114 INVITE
Server: Asterisk PBX 1.6.1.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Apr 21 08:01:16] NOTICE[11906]: chan_sip.c:18527
handle_request_invite: Call from '' to extension '331NUMNOFOUND'
rejected because extension not found.




a idea of the problems ?

My supplier use a lot of server, i thinkss that my asterisk don't link
IP of the incoming server to the extensions


thanks for your help
olivier



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