Hi Oliver , This is a simple scenario with asterisk you can edit sip.conf and in peer entry, try to add, context=(desired_context for peer)
and then into context write a dial-plan for given number and route a call or whatever you want to do. On Wed, Mar 23, 2011 at 11:31 AM, Olivier CALVANO <[email protected]>wrote: > Hi > > I request your help because i don't have actually a solution at my > problems. > > > I have a Asterisk Server in 1.6 > Connected at a SIP Provider > This provider supply me 2 numbers: > 003318364xxxx (official number) > 081169xxxx (Nddi Number) > > When i receive a call on the 081169xxxx, he don't use > the extension. He use the 003318364xxxx extension. > > SIP Debug: > > <--- SIP read from UDP://91.121.xxx.xxx:5060 ---> > INVITE sip:[email protected]:5060;transport=udp SIP/2.0 > Allow: UPDATE,REFER,INFO > Call-ID: [email protected] > Contact: <sip:91.121.xxx.xxx:5060> > Content-Type: application/sdp > CSeq: 1602837515 INVITE > From: "033426aaaaaa" > <sip:[email protected] > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > Max-Forwards: 30 > P-Preferred-Identity: <sip:[email protected];user=phone> > To: <sip:[email protected];user=phone> > User-Agent: Cirpack/v4.42s (gw_sip) > Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 > Content-Length: 481 > > v=0 > o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 > s=SIP Call > c=IN IP4 91.121.bbb.bbb > t=0 0 > m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 > b=AS:21 > a=rtpmap:18 G729/8000/1 > a=fmtp:18 annexb=no > a=rtpmap:4 G723/8000/1 > a=fmtp:4 annexa=no > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:125 CLEARMODE/8000/1 > a=rtpmap:111 iLBC/8000/1 > a=fmtp:111 mode=30 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:30 > a=sendrecv > a=sqn:0 > a=cdsc: 1 image udptl t38 > > <-------------> > --- (13 headers 22 lines) --- > Sending to 91.121.xxx.xxx : 5060 (no NAT) > Using INVITE request as basis request - > [email protected] > Found peer 'Myoperator' for '033426aaaaaa' from 91.121.xxx.xxx:5060 > Found RTP audio format 18 > Found RTP audio format 4 > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 125 > Found RTP audio format 111 > Found RTP audio format 101 > Peer audio RTP is at port 91.121.bbb.bbb:36146 > 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 audio description format PCMA for ID 8 > Found unknown media description format CLEARMODE for ID 125 > Found audio description format iLBC for ID 111 > Found audio description format telephone-event for ID 101 > Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d > (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), > combined - 0x109 (g723|alaw|g729) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 91.121.bbb.bbb:36146 > Looking for 003318364xxxx in Appels-Entrants (domain 78.41.xxx.xxx) > > <--- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 ---> > SIP/2.0 404 Not Found > Via: SIP/2.0/UDP > 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx > From: "033426aaaaaa" > <sip:[email protected] > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > To: <sip:[email protected];user=phone>;tag=as50e04b6a > Call-ID: [email protected] > CSeq: 1602837515 INVITE > Server: Asterisk PBX 1.6.1.8 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Length: 0 > > > <------------> > [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 > handle_request_invite: Call from '0033459aaaaaa' to extension > '003318364xxxx' rejected because extension not found. > Scheduling destruction of SIP dialog > '[email protected]' in 6400 ms (Method: > INVITE) > <--- SIP read from UDP://91.121.xxx.xxx:5060 ---> > ACK sip:[email protected]:5060;transport=udp SIP/2.0 > Call-ID: [email protected] > Contact: <sip:91.121.xxx.xxx:5060> > CSeq: 1602837515 ACK > From: "033426aaaaaa" > <sip:[email protected] > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > Max-Forwards: 30 > To: <sip:[email protected];user=phone>;tag=as50e04b6a > User-Agent: Cirpack/v4.42s (gw_sip) > Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 > Content-Length: 0 > > > > > > > > I see in the debug: > To: <sip:[email protected];user=phone> > > but he search the 003318364xxxx extension > [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 > handle_request_invite: Call from '0033459aaaaaa' to extension > '003318364xxxx' rejected because extension not found. > > > > > Anyone know the solution for he use the extension based on the "To:" ? > > thanks > Olivier > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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