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Sip configuration


roger
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I Got a CS410, I have it configured to register againt a SIP trunk, every thing work normal, the PBX regiter fine... all good.

 

when I call the Trunk ID number Example 317-8888888 it said you have reached a non working number, but If I go to the Trunk configuartion tab and scroll down to the option foward all call to and put and extesion number The PBX will forwad the incoming call to the assigned extension.

 

Is like the PBX do not know what to do with the incoming call, Yes I have assigned the Number to the AA.

 

I have traced the call and it arriving to the PBX but it not going any were.

 

 

Please Help.

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I Got a CS410, I have it configured to register againt a SIP trunk, every thing work normal, the PBX regiter fine... all good.

 

when I call the Trunk ID number Example 317-8888888 it said you have reached a non working number, but If I go to the Trunk configuartion tab and scroll down to the option foward all call to and put and extesion number The PBX will forwad the incoming call to the assigned extension.

 

Is like the PBX do not know what to do with the incoming call, Yes I have assigned the Number to the AA.

 

I have traced the call and it arriving to the PBX but it not going any were.

 

 

Please Help.

 

What firmware version?

Did you try adding the phone number as an alias on the auto attendant?

If you put the extension number of the auto attendant in the forward all calls it works?

Can you paste the log of an incoming call to the forum?

Are you using an ITSP? if yes which one?

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What firmware version?

Did you try adding the phone number as an alias on the auto attendant?

If you put the extension number of the auto attendant in the forward all calls it works?

Can you paste the log of an incoming call to the forum?

Are you using an ITSP? if yes which one?

Yes I have added the number as an alias and it dont work.

Yes it will forward all call to that extension.

the log is below.

Egix.

 

thanks

 

 

 

 

[7] 2008/12/16 21:23:01: SIP Rx udp:209.131.220.220:62328:

INVITE sip:egix@66.158.175.178:5060;line=a87ff679;transport=udp SIP/2.0

Via:SIP/2.0/UDP 209.131.220.220;branch=z9hG4bK-BroadWorks.as01-66.158.175.178V5060-0-559967328-308845606-1229480581310-

From:"ROGER WATSON CA"<sip:31734218@209.131.220.220;user=phone>;tag=308845606-1229480581310-

To:"3176604804 3176604804"<sip:3176604804@egix.net;line=a87ff679>

Call-ID:BW212301310161208-520487878@209.131.220.220

CSeq:559967328 INVITE

Contact:<sip:209.131.220.220:5060>

Supported:100rel

Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE

Accept:multipart/mixed,application/media_control+xml,application/sdp

Max-Forwards:10

Content-Type:application/sdp

Content-Length:281

 

v=0

o=BroadWorks 77316568 1 IN IP4 209.131.222.201

s=-

c=IN IP4 209.131.222.201

t=0 0

m=audio 16574 RTP/AVP 0 100 101

c=IN IP4 209.131.xxx.201

a=rtpmap:0 PCMU/8000

a=rtpmap:100 X-NSE/8000

a=fmtp:100 192-194

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

[5] 2008/12/16 21:23:01: Identify trunk (line match) 4

[9] 2008/12/16 21:23:01: Resolve 180: aaaa udp 209.131.xxx.220 5060

[9] 2008/12/16 21:23:01: Resolve 180: a udp 209.131.220.xxx 5060

[9] 2008/12/16 21:23:01: Resolve 180: udp 209.131.220.xxx 5060

[7] 2008/12/16 21:23:01: SIP Tx udp:209.131.xxx.220:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 209.131.xxx.220;branch=z9hG4bK-BroadWorks.as01-66.158.175.178V5060-0-559967328-308845606-1229480581310-

From: "ROGER WATSON CA" <sip:3178334218@209.131.xxx.220;user=phone>;tag=308845606-1229480581310-

To: "3176604804 3176604804" <sip:3176604804@egix.net;line=a87ff679>;tag=5706a1ee13

Call-ID: BW212301310161208-520487878@209.131.220.220

CSeq: 559967328 INVITE

Content-Length: 0

 

 

[5] 2008/12/16 21:23:01: Trunk 3176604804 sends call to egix in domain localhost

[5] 2008/12/16 21:23:01: Trunk call: Could not identify user

[9] 2008/12/16 21:23:01: Resolve 181: aaaa udp 209.131.220.220 5060

[9] 2008/12/16 21:23:01: Resolve 181: a udp 209.131.220.220 5060

[9] 2008/12/16 21:23:01: Resolve 181: udp 209.131.220.220 5060

[7] 2008/12/16 21:23:01: SIP Tx udp:209.131.220.220:5060:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 209.131.220.220;branch=z9hG4bK-BroadWorks.as01-66.158.xxx.178V5060-0-559967328-308845606-1229480581310-

From: "ROGER WATSON CA" <sip:3178334218@209.131.220.220;user=phone>;tag=308845606-1229480581310-

To: "3176604804 3176604804" <sip:3176604804@egix.net;line=a87ff679>;tag=5706a1ee13

Call-ID: BW212301310161208-520487878@209.131.220.220

CSeq: 559967328 INVITE

Contact: <sip:egix@66.158.xxx.178:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.0.1.3023

Content-Length: 0

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[...]

INVITE sip:egix@66.158.175.178:5060;line=a87ff679;transport=udp SIP/2.0

[...]

[5] 2008/12/16 21:23:01: Trunk 3176604804 sends call to egix in domain localhost

[5] 2008/12/16 21:23:01: Trunk call: Could not identify user

 

In SIP the destination is in the first line (the Request-URI, see http://wiki.pbxnsip.com/index.php/Request-URI). That means the PBX is searching for the user "egix".

 

If you want to use the "To"-header, then you can use the following pattern in the setting "Send call to extension": !(.*)!\1!t!

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