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

Sun Mar 27 23:20:49 CDT 2011

Hi


Very thanks for your helps, that's work very goo

Bye
Olivier


2011/3/25 DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>:
> 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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>
> --
> _____________________________________________________________________
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