[asterisk-users] sipgate outgoing calls

Thu Sep 19 05:09:19 CDT 2013

Challenge authentication look good.

<--- SIP read from UDP:217.10.79.23:5060 --->
SIP/2.0 200 OK

Are you sure this number format  01179553708 is accepted in that SIP trunk?
Some VOIP providers only accept international format.

-- 
==================================
Miguel Oyarzo
DevOps Engineer
http://www.linkedin.com/in/mikeaustralia
Linux User: # 483188 - counter.li.org
Melbourne, Australia

On 9/19/2013 7:55 PM, gpxctawjc5oh at irational.org wrote:
>> It looks like the challenge response after INVITE is not been accepted.
>>
>> Provide more detail.
>>
>> $> sip set debug peer sipgate
>
> server*CLI> sip set debug peer sipgate
> SIP Debugging Enabled for IP: 217.10.79.23:5060
> Really destroying SIP dialog 
> '3ef8ff1a6ec360626af409b112b860ee at 127.0.1.1' Method: REGISTER
>     -- Registered SIP 'xxxxx' at 86.140.115.135 port 5060
>   == Using SIP RTP CoS mark 5
>   == Using SIP VRTP CoS mark 6
>
>
>     -- Executing [01179553708 at default:1] Set("SIP/xxxxx-0000015d", 
> "CALLERID(num)=xxxxx") in new stack
>     -- Executing [01179553708 at default:2] Dial("SIP/xxxxx-0000015d", 
> "SIP/01179553708 at sipgate,30,trg") in new stack
>   == Using SIP RTP CoS mark 5
>   == Using SIP VRTP CoS mark 6
>     -- Called 01179553708 at sipgate
> [Sep 19 10:50:15] NOTICE[28232]: chan_sip.c:17885 
> handle_response_invite: Failed to authenticate on INVITE to '"xxxxx" 
> <sip:xxxxx at sipgate.co.uk>;tag=as629ee6f8'
>     -- SIP/sipgate-0000015e is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
>     -- Executing [01179553708 at default:3] Hangup("SIP/xxxxx-0000015d", 
> "") in new stack
>   == Spawn extension (default, 01179553708, 3) exited non-zero on 
> 'SIP/xxxxx-0000015d'
>
>
> ---
> server*CLI>
> <--- SIP read from UDP:217.10.79.23:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK73a1638c;rport=5060
> From: "asterisk" <sip:asterisk at 92.63.131.3>;tag=as5dcb32d8
> To: <sip:sipgate.co.uk>;tag=99199803810c7e807ea44745826d9aa4.df2d
> Call-ID: 65cb07675eefaaef5f655e8a0be6b2f6 at 92.63.131.3
> CSeq: 102 OPTIONS
> Accept: */*
> Accept-Encoding:
> Accept-Language: en
> Supported:
> Content-Length: 0
>
>
> <------------->
> --- (11 headers 0 lines) ---
> Really destroying SIP dialog 
> '65cb07675eefaaef5f655e8a0be6b2f6 at 92.63.131.3' Method: OPTIONS
> [Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:11588 sip_reregister:    
> -- Re-registration for  xxxxxx at sipgate.co.uk
> REGISTER 12 headers, 0 lines
> Reliably Transmitting (no NAT) to 217.10.79.23:5060:
> REGISTER sip:sipgate.co.uk SIP/2.0
> Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK13b202d7;rport
> Max-Forwards: 70
> From: <sip:xxxx at sipgate.co.uk>;tag=as19513575
> To: <sip:xxxxx at sipgate.co.uk>
> Call-ID: 3ef8ff1a6ec360626af409b112b860ee at 127.0.1.1
> CSeq: 182 REGISTER
> User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
> Authorization: Digest username="xxxxxx", realm="sipgate.co.uk", 
> algorithm=MD5, uri="sip:sipgate.co.uk", 
> nonce="523ac9531b1cc7962e07bce6a76683ee24da44d0", 
> response="c82fac231a41085c275899ad84f73317"
> Expires: 120
> Contact: <sip:xxxxxx at 92.63.131.3>
> Content-Length: 0
>
>
> ---
> server*CLI>
> <--- SIP read from UDP:217.10.79.23:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 
> 92.63.131.3:5060;received=92.63.131.3;branch=z9hG4bK13b202d7;rport=5060
> From: <sip:xxxxx at sipgate.co.uk>;tag=as19513575
> To: <sip:xxxxx at sipgate.co.uk>;tag=c3e497ecaece77a8e244e564b4212178.3e46
> Call-ID: 3ef8ff1a6ec360626af409b112b860ee at 127.0.1.1
> CSeq: 182 REGISTER
> Contact: <sip:xxxxxx at 92.63.131.3>;expires=120
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Scheduling destruction of SIP dialog 
> '3ef8ff1a6ec360626af409b112b860ee at 127.0.1.1' in 32000 ms (Method: 
> REGISTER)
> [Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:18301 
> handle_response_register: Outbound Registration: Expiry for 
> sipgate.co.uk is 120 sec (Scheduling reregistration in 105 s)
> Reliably Transmitting (no NAT) to 217.10.79.23:5060:
> OPTIONS sip:sipgate.co.uk SIP/2.0
> Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk at 92.63.131.3>;tag=as5afd24b2
> To: <sip:sipgate.co.uk>
> Contact: <sip:asterisk at 92.63.131.3>
> Call-ID: 1becd4dc336869c4692fc4e55e109562 at 92.63.131.3
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4
> Date: Thu, 19 Sep 2013 09:51:27 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Content-Length: 0
>
>
> ---
> server*CLI>
> <--- SIP read from UDP:217.10.79.23:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport=5060
> From: "asterisk" <sip:asterisk at 92.63.131.3>;tag=as5afd24b2
> To: <sip:sipgate.co.uk>;tag=99199803810c7e807ea44745826d9aa4.c753
> Call-ID: 1becd4dc336869c4692fc4e55e109562 at 92.63.131.3
> CSeq: 102 OPTIONS
> Accept: */*
> Accept-Encoding:
> Accept-Language: en
> Supported:
> Content-Length: 0
>
>
> <------------->
> --- (11 headers 0 lines) ---
> Really destroying SIP dialog 
> '1becd4dc336869c4692fc4e55e109562 at 92.63.131.3' Method: OPTIONS
>
> -- 
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