Hi
Very thanks for your helps, that's work very goo Bye Olivier 2011/3/25 DHAVAL INDRODIYA <[email protected]>: > 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 <[email protected]> > wrote: >> >> Hi >> >> Anyone know a solution at my problems ? >> >> Thanks >> Olivier >> >> >> >> >> >> >> >> 2011/3/23 Olivier CALVANO <[email protected]>: >> > 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 > > > -- > _____________________________________________________________________ > -- 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 > -- _____________________________________________________________________ -- 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
