Hi Olivier, here is solutions for your situation , ideally you need to talk with Provider and they can set SIP URI for given DID numbre , but that can be solved by dial-plan like this.
exten => _003318364xxxx,1,Set(foo=${SIP_HEADER(To)}) exten => _003318364xxxx,n,Set(cut1=${CUT(foo,:,2)}) exten => _003318364xxxx,n,Set(CLI=${CUT(cut1,>,1)}) exten => _003318364xxxx,n,Set(toexten=${CUT(CLI,@,1)}) exten => _003318364xxxx,n,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten => _003318364xxxx,n,ExecIf($["${toexten}" = "81169xxxx"]?Dial(SIP/204,180,rt):Noop(${toexten})) exten => _003318364xxxx,n,ExecIf($["${EXTEN}" = "003318364xxxx"]?Dial(SIP/203,180,rt):Noop(${toexten})) On Thu, Mar 24, 2011 at 11:13 AM, Olivier CALVANO <o.calv...@gmail.com>wrote: > Hi > > Anyone know a solution at my problems ? > > Thanks > Olivier > > > > > > > > 2011/3/23 Olivier CALVANO <o.calv...@gmail.com>: > > 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:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 > > Allow: UPDATE,REFER,INFO > > Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net > > Contact: <sip:91.121.xxx.xxx:5060> > > Content-Type: application/sdp > > CSeq: 1602837515 INVITE > > From: "033426aaaaaa" > > <sip:033426aaa...@sip.myoperator.net > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > > Max-Forwards: 30 > > P-Preferred-Identity: <sip:033426aaa...@sip.myoperator.net;user=phone> > > To: <sip:081169x...@91.121.xxx.xxx;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 - > > 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net > > 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:033426aaa...@sip.myoperator.net > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > > To: <sip:081169x...@91.121.xxx.xxx;user=phone>;tag=as50e04b6a > > Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net > > 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 > > '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method: > > INVITE) > > <--- SIP read from UDP://91.121.xxx.xxx:5060 ---> > > ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 > > Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net > > Contact: <sip:91.121.xxx.xxx:5060> > > CSeq: 1602837515 ACK > > From: "033426aaaaaa" > > <sip:033426aaa...@sip.myoperator.net > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > > Max-Forwards: 30 > > To: <sip:081169x...@91.121.xxx.xxx;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:081169x...@91.121.xxx.xxx;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? 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-- _____________________________________________________________________ -- 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