Re: [asterisk-users] sipgate outgoing calls

Miguel Oyarzo Thu, 19 Sep 2013 03:09:45 -0700


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, [email protected] 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'[email protected]' 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@default:1] Set("SIP/xxxxx-0000015d","CALLERID(num)=xxxxx") in new stack-- Executing [01179553708@default:2] Dial("SIP/xxxxx-0000015d","SIP/01179553708@sipgate,30,trg") in new stack

  == Using SIP RTP CoS mark 5
  == Using SIP VRTP CoS mark 6
    -- Called 01179553708@sipgate

[Sep 19 10:50:15] NOTICE[28232]: chan_sip.c:17885handle_response_invite: Failed to authenticate on INVITE to '"xxxxx"<sip:[email protected]>;tag=as629ee6f8'

    -- SIP/sipgate-0000015e is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

-- Executing [01179553708@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:[email protected]>;tag=as5dcb32d8
To: <sip:sipgate.co.uk>;tag=99199803810c7e807ea44745826d9aa4.df2d
Call-ID: [email protected]
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

Really destroying SIP dialog'[email protected]' Method: OPTIONS[Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:11588 sip_reregister:-- Re-registration for [email protected]

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:[email protected]>;tag=as19513575
To: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>
Content-Length: 0


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

Via: SIP/2.0/UDP92.63.131.3:5060;received=92.63.131.3;branch=z9hG4bK13b202d7;rport=5060

From: <sip:[email protected]>;tag=as19513575
To: <sip:[email protected]>;tag=c3e497ecaece77a8e244e564b4212178.3e46
Call-ID: [email protected]
CSeq: 182 REGISTER
Contact: <sip:[email protected]>;expires=120
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

Scheduling destruction of SIP dialog'[email protected]' in 32000 ms (Method:REGISTER)[Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:18301handle_response_register: Outbound Registration: Expiry forsipgate.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:[email protected]>;tag=as5afd24b2
To: <sip:sipgate.co.uk>
Contact: <sip:[email protected]>
Call-ID: [email protected]
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:[email protected]>;tag=as5afd24b2
To: <sip:sipgate.co.uk>;tag=99199803810c7e807ea44745826d9aa4.c753
Call-ID: [email protected]
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---

Really destroying SIP dialog'[email protected]' Method: OPTIONS


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Re: [asterisk-users] sipgate outgoing calls

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