[asterisk-users] No extension found ?

Mon Apr 23 00:57:58 CDT 2012

Hi,

No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'

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)

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.

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.!!

Regards,
Sammy.

On Sat, Apr 21, 2012 at 11:47 AM, Michel Verbraak <michel at verbraak.org>wrote:

> On 21-04-12 08:19, Olivier CALVANO wrote:
>
>> 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=domaineofmysuppl**ier.net <http://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=**
>> z9hG4bK00151747E2606DB6CA39464**AF542
>> From: "+331MYCLID"
>> <sip:+331MYCLID;tgrp=RT43 at 172.**16.21.11 <RT43 at 172.16.21.11>>;tag=**
>> 2RUVP51HBW30000E1D00001u0K4NFQ**C0QNAN31
>> To:<sip:+331NUMNOFOUND at 172.16.**20.119<sip%3A%2B331NUMNOFOUND at 172.16.20.119>
>> >
>> Call-ID: 60471500e217-4f924d2c-**477df10c-66ea6f8-140732f at 127.**0.0.1<60471500e217-4f924d2c-477df10c-66ea6f8-140732f at 127.0.0.1>
>> CSeq: 20114 INVITE
>> Contact:<sip:+331MYCLID at 172.**16.21.11:5060<http://sip:+331MYCLID@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<sip%3A%2B331MYCLID at domaineofmysupplier.net>
>> >
>> Supported: timer, replaces
>> Content-Length: 369
>> Min-SE: 90
>> Session-Expires: 300
>> P-Charging-Vector: icid-value="4f924d2c1e20abe1d@**172.16.20.119<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<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)
>>
> It is looking for the 331NUMNOFOUND in context named "default".
> Do you have this context? Does the extension exists in the context?
>
> 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.
>
> For example: register => username:password at sip.**
> voipbuster.com/Trunk-Telco<http://username:password@sip.voipbuster.com/Trunk-Telco>
>
>
>> <--- 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
>>
> <snip>
>
>  <------------>
>> [Apr 21 08:01:16] NOTICE[11906]: chan_sip.c:18527
>> handle_request_invite: Call from '' to extension '331NUMNOFOUND'
>> rejected because extension not found.
>>
>>
>>
>>
>>
>>  Regards,
> Michel.
>
>
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