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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

head link

[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 --------------
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Olivier CALVANO

2011-Mar-24 05:43 UTC

head link

[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
>

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