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

Wed Mar 23 03:37:17 CDT 2011

Hi Dhaval,

Thanks for your answer, but i not my question ;=)

My asterisk have a entry into the sip.conf with a context.

in extensions.conf, i have this extensions:

        exten => _003318364xxxx,1,Dial(SIP/203,180,rt)
        exten => _003381169xxxx,1,Dial(SIP/204,180,rt)

(in my debug, i have deleted the exten => _003318364xxxx)

When i call to 3318364xxxx that's work
When i call to 3381169xxxx that's work but it's the _003318364xxxx is
used and phone 203 ring

bye
olivier

2011/3/23 DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>:
> 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
>>
>> --
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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
>
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