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=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:[email protected]: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;[email protected]>;tag=2RUVP51HBW30000E1D00001u0K4NFQC0QNAN31
To:<sip:[email protected]>
Call-ID: [email protected]
CSeq: 20114 INVITE
Contact:<sip:[email protected]: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:[email protected]>
Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value="[email protected]"
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 -
[email protected]
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:[email protected]/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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