Hi Oliver ,<br><br>This is a simple scenario with asterisk you can edit sip.conf and in peer entry, try to add,<br>context=(desired_context for peer)<br><br>and then into context write a dial-plan for given number and route a call or whatever you want to do.<br>
<br><div class="gmail_quote">On Wed, Mar 23, 2011 at 11:31 AM, Olivier CALVANO <span dir="ltr"><<a href="mailto:o.calvano@gmail.com">o.calvano@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi<br>
<br>
I request your help because i don't have actually a solution at my problems.<br>
<br>
<br>
I have a Asterisk Server in 1.6<br>
Connected at a SIP Provider<br>
This provider supply me 2 numbers:<br>
003318364xxxx (official number)<br>
081169xxxx (Nddi Number)<br>
<br>
When i receive a call on the 081169xxxx, he don't use<br>
the extension. He use the 003318364xxxx extension.<br>
<br>
SIP Debug:<br>
<br>
<--- SIP read from UDP://91.121.xxx.xxx:5060 ---><br>
INVITE sip:003318364xxxx@78.41.xxx.xxx:5060;transport=udp SIP/2.0<br>
Allow: UPDATE,REFER,INFO<br>
Call-ID: <a href="mailto:04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net">04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net</a><br>
Contact: <sip:91.121.xxx.xxx:5060><br>
Content-Type: application/sdp<br>
CSeq: 1602837515 INVITE<br>
From: "033426aaaaaa"<br>
<<a href="mailto:sip%3A033426aaaaaa@sip.myoperator.net">sip:033426aaaaaa@sip.myoperator.net</a>;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60<br>
Max-Forwards: 30<br>
P-Preferred-Identity: <<a href="mailto:sip%3A033426aaaaaa@sip.myoperator.net">sip:033426aaaaaa@sip.myoperator.net</a>;user=phone><br>
To: <sip:081169xxxx@91.121.xxx.xxx;user=phone><br>
User-Agent: Cirpack/v4.42s (gw_sip)<br>
Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2<br>
Content-Length: 481<br>
<br>
v=0<br>
o=cp10 130085910854 130085910854 IN IP4 10.7.1.121<br>
s=SIP Call<br>
c=IN IP4 91.121.bbb.bbb<br>
t=0 0<br>
m=audio 36146 RTP/AVP 18 4 0 8 125 111 101<br>
b=AS:21<br>
a=rtpmap:18 G729/8000/1<br>
a=fmtp:18 annexb=no<br>
a=rtpmap:4 G723/8000/1<br>
a=fmtp:4 annexa=no<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:8 PCMA/8000/1<br>
a=rtpmap:125 CLEARMODE/8000/1<br>
a=rtpmap:111 iLBC/8000/1<br>
a=fmtp:111 mode=30<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=ptime:30<br>
a=sendrecv<br>
a=sqn:0<br>
a=cdsc: 1 image udptl t38<br>
<br>
<-------------><br>
--- (13 headers 22 lines) ---<br>
Sending to 91.121.xxx.xxx : 5060 (no NAT)<br>
Using INVITE request as basis request -<br>
<a href="mailto:04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net">04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net</a><br>
Found peer 'Myoperator' for '033426aaaaaa' from 91.121.xxx.xxx:5060<br>
Found RTP audio format 18<br>
Found RTP audio format 4<br>
Found RTP audio format 0<br>
Found RTP audio format 8<br>
Found RTP audio format 125<br>
Found RTP audio format 111<br>
Found RTP audio format 101<br>
Peer audio RTP is at port 91.121.bbb.bbb:36146<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 audio description format PCMA for ID 8<br>
Found unknown media description format CLEARMODE for ID 125<br>
Found audio description format iLBC for ID 111<br>
Found audio description format telephone-event for ID 101<br>
Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d<br>
(g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing),<br>
combined - 0x109 (g723|alaw|g729)<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 91.121.bbb.bbb:36146<br>
Looking for 003318364xxxx in Appels-Entrants (domain 78.41.xxx.xxx)<br>
<br>
<--- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 ---><br>
SIP/2.0 404 Not Found<br>
Via: SIP/2.0/UDP<br>
91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx<br>
From: "033426aaaaaa"<br>
<<a href="mailto:sip%3A033426aaaaaa@sip.myoperator.net">sip:033426aaaaaa@sip.myoperator.net</a>;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60<br>
To: <sip:081169xxxx@91.121.xxx.xxx;user=phone>;tag=as50e04b6a<br>
Call-ID: <a href="mailto:04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net">04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net</a><br>
CSeq: 1602837515 INVITE<br>
Server: Asterisk PBX 1.6.1.8<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces, timer<br>
Content-Length: 0<br>
<br>
<br>
<------------><br>
[Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527<br>
handle_request_invite: Call from '0033459aaaaaa' to extension<br>
'003318364xxxx' rejected because extension not found.<br>
Scheduling destruction of SIP dialog<br>
'<a href="mailto:04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net">04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net</a>' in 6400 ms (Method:<br>
INVITE)<br>
<--- SIP read from UDP://91.121.xxx.xxx:5060 ---><br>
ACK sip:003318364xxxx@78.41.xxx.xxx:5060;transport=udp SIP/2.0<br>
Call-ID: <a href="mailto:04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net">04459-NK-5fa6f8a0-18641fd41@sip.myoperator.net</a><br>
Contact: <sip:91.121.xxx.xxx:5060><br>
CSeq: 1602837515 ACK<br>
From: "033426aaaaaa"<br>
<<a href="mailto:sip%3A033426aaaaaa@sip.myoperator.net">sip:033426aaaaaa@sip.myoperator.net</a>;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60<br>
Max-Forwards: 30<br>
To: <sip:081169xxxx@91.121.xxx.xxx;user=phone>;tag=as50e04b6a<br>
User-Agent: Cirpack/v4.42s (gw_sip)<br>
Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2<br>
Content-Length: 0<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
I see in the debug:<br>
To: <sip:081169xxxx@91.121.xxx.xxx;user=phone><br>
<br>
but he search the 003318364xxxx extension<br>
[Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527<br>
handle_request_invite: Call from '0033459aaaaaa' to extension<br>
'003318364xxxx' rejected because extension not found.<br>
<br>
<br>
<br>
<br>
Anyone know the solution for he use the extension based on the "To:" ?<br>
<br>
thanks<br>
Olivier<br>
<br>
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</blockquote></div><br>