[asterisk-users] Problems Extension with a Call In on Asterisk 1.6

[DHAVAL INDRODIYA]

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.calvano at gmail.com>wrote:

> 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
> >
>
> --
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