Olivier CALVANO
2011-Mar-23 06:01 UTC
[asterisk-users] Problems Extension with a Call In on Asterisk 1.6
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:003318364xxxx at 78.41.xxx.xxx:5060;transport=udp SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 04459-NK-5fa6f8a0-18641fd41 at sip.myoperator.net Contact: <sip:91.121.xxx.xxx:5060> Content-Type: application/sdp CSeq: 1602837515 INVITE From: "033426aaaaaa" <sip:033426aaaaaa at sip.myoperator.net;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: <sip:033426aaaaaa at sip.myoperator.net;user=phone> To: <sip:081169xxxx at 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-5fa6f8a0-18641fd41 at 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:033426aaaaaa at sip.myoperator.net;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 To: <sip:081169xxxx at 91.121.xxx.xxx;user=phone>;tag=as50e04b6a Call-ID: 04459-NK-5fa6f8a0-18641fd41 at 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-5fa6f8a0-18641fd41 at sip.myoperator.net' in 6400 ms (Method: INVITE) <--- SIP read from UDP://91.121.xxx.xxx:5060 ---> ACK sip:003318364xxxx at 78.41.xxx.xxx:5060;transport=udp SIP/2.0 Call-ID: 04459-NK-5fa6f8a0-18641fd41 at sip.myoperator.net Contact: <sip:91.121.xxx.xxx:5060> CSeq: 1602837515 ACK From: "033426aaaaaa" <sip:033426aaaaaa at sip.myoperator.net;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 To: <sip:081169xxxx at 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:081169xxxx at 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
DHAVAL INDRODIYA
2011-Mar-23 07:05 UTC
[asterisk-users] Problems Extension with a Call In on Asterisk 1.6
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 <o.calvano at gmail.com>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:003318364xxxx at 78.41.xxx.xxx:5060;transport=udp SIP/2.0 > Allow: UPDATE,REFER,INFO > Call-ID: 04459-NK-5fa6f8a0-18641fd41 at sip.myoperator.net > Contact: <sip:91.121.xxx.xxx:5060> > Content-Type: application/sdp > CSeq: 1602837515 INVITE > From: "033426aaaaaa" > <sip:033426aaaaaa at sip.myoperator.net > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > Max-Forwards: 30 > P-Preferred-Identity: <sip:033426aaaaaa at sip.myoperator.net;user=phone> > To: <sip:081169xxxx at 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-5fa6f8a0-18641fd41 at 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:033426aaaaaa at sip.myoperator.net > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > To: <sip:081169xxxx at 91.121.xxx.xxx;user=phone>;tag=as50e04b6a > Call-ID: 04459-NK-5fa6f8a0-18641fd41 at 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-5fa6f8a0-18641fd41 at sip.myoperator.net' in 6400 ms (Method: > INVITE) > <--- SIP read from UDP://91.121.xxx.xxx:5060 ---> > ACK sip:003318364xxxx at 78.41.xxx.xxx:5060;transport=udp SIP/2.0 > Call-ID: 04459-NK-5fa6f8a0-18641fd41 at sip.myoperator.net > Contact: <sip:91.121.xxx.xxx:5060> > CSeq: 1602837515 ACK > From: "033426aaaaaa" > <sip:033426aaaaaa at sip.myoperator.net > ;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > Max-Forwards: 30 > To: <sip:081169xxxx at 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:081169xxxx at 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? 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110323/fa0d0c67/attachment.htm>
Olivier CALVANO
2011-Mar-24 05:43 UTC
[asterisk-users] Problems Extension with a Call In on Asterisk 1.6
Hi Anyone know a solution at my problems ? Thanks Olivier 2011/3/23 Olivier CALVANO <o.calvano at 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:003318364xxxx at 78.41.xxx.xxx:5060;transport=udp SIP/2.0 > Allow: UPDATE,REFER,INFO > Call-ID: 04459-NK-5fa6f8a0-18641fd41 at sip.myoperator.net > Contact: <sip:91.121.xxx.xxx:5060> > Content-Type: application/sdp > CSeq: 1602837515 INVITE > From: "033426aaaaaa" > <sip:033426aaaaaa at sip.myoperator.net;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > Max-Forwards: 30 > P-Preferred-Identity: <sip:033426aaaaaa at sip.myoperator.net;user=phone> > To: <sip:081169xxxx at 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-5fa6f8a0-18641fd41 at 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:033426aaaaaa at sip.myoperator.net;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > To: <sip:081169xxxx at 91.121.xxx.xxx;user=phone>;tag=as50e04b6a > Call-ID: 04459-NK-5fa6f8a0-18641fd41 at 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-5fa6f8a0-18641fd41 at sip.myoperator.net' in 6400 ms (Method: > INVITE) > <--- SIP read from UDP://91.121.xxx.xxx:5060 ---> > ACK sip:003318364xxxx at 78.41.xxx.xxx:5060;transport=udp SIP/2.0 > Call-ID: 04459-NK-5fa6f8a0-18641fd41 at sip.myoperator.net > Contact: <sip:91.121.xxx.xxx:5060> > CSeq: 1602837515 ACK > From: "033426aaaaaa" > <sip:033426aaaaaa at sip.myoperator.net;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 > Max-Forwards: 30 > To: <sip:081169xxxx at 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:081169xxxx at 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 >