[Asterisk-Users] Asterisk as SIP Proxy

[Olle E. Johansson]

John Todd wrote:

> Perhaps it's because the Contact: field does not have an extension in 
> it, just an IP address?  This is a guess without really thinking about 
> it too much.
The Contact: field sure looks weird, but the SIPDOMAIN comes from
the INVITE - or?

Ranga, please check your debug log in /var/log/asterisk too see if
the SIP channel chokes on the Contact: field and gives up parsing.
Maybe there's an error message in there. Just guessing here.

/O

> 
>> Ranga,
>> I'm sorry, I can't find the error in this configuration. I called on 
>> IP address myself,
>> and my Asterisk picked out the IP address into the domain part and 
>> dialed out.
>>
>> I'm stuck. Anyone else that see the problem?
>>
>> /O
>>
>> ranga wrote:
>>
>>> Here it goes
>>>
>>>
>>> Sip read: CLI>
>>> INVITE sip:ranga at 192.168.68.6 SIP/2.0
>>> Content-Length: 116
>>> Contact: <sip:192.168.68.12>
>>> Call-ID: F1C6B16E-51A7-4987-A2B9-128C2302378C at 192.168.68.12
>>> Content-Type: application/sdp
>>> Max-Forwards: 70
>>> From: "Ranga Rao Vutukuru"<sip:sridhar at 192.168.68.15>;tag=21632105
>>> CSeq: 1 INVITE
>>> To: <sip:ranga at 192.168.68.6>
>>> Via: SIP/2.0/UDP 192.168.68.12:5060
>>>
>>> v=0
>>> o=- 3279257833 3279257833 IN IP4 192.168.68.12
>>> s=-
>>> c=IN IP4 192.168.68.12
>>> t=0 0
>>> m=audio 16390 RTP/AVP 8 0
>>>
>>> 10 headers, 6 lines
>>> Using latest request as basis request
>>> Sending to 192.168.68.12 : 5060 (non-NAT)
>>> Found audio format ALAW
>>> Found audio format UNKN
>>> Capabilities: us - 524302, them - 12/0, combined - 12
>>> Non-codec capabilities: us - 1, them - 0, combined - 0
>>> Reliably Transmitting (no NAT):
>>> SIP/2.0 407 Proxy Authentication Required
>>> Via: SIP/2.0/UDP 192.168.68.12:5060
>>> From: "Ranga Rao Vutukuru"<sip:sridhar at 192.168.68.15>;tag=21632105
>>> To: <sip:ranga at 192.168.68.6>;tag=as78933dd8
>>> Call-ID: F1C6B16E-51A7-4987-A2B9-128C2302378C at 192.168.68.12
>>> CSeq: 1 INVITE
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>>> Contact:
>>> Proxy-Authenticate: Digest realm="asterisk", nonce="25230b01"
>>> Content-Length: 0
>>>
>>>
>>>  to 192.168.68.12:5060
>>> Sip read: CLI>
>>> ACK sip:ranga at 192.168.68.6 SIP/2.0
>>> Content-Length: 0
>>> Call-ID: F1C6B16E-51A7-4987-A2B9-128C2302378C at 192.168.68.12
>>> CSeq: 1 ACK
>>> From: "Ranga Rao Vutukuru"<sip:sridhar at 192.168.68.15>;tag=21632105
>>> To: <sip:ranga at 192.168.68.6>;tag=as78933dd8
>>> Via: SIP/2.0/UDP 192.168.68.12:5060
>>>
>>>
>>> 7 headers, 0 lines
>>> Sip read: CLI>
>>> INVITE sip:ranga at 192.168.68.6 SIP/2.0
>>> Content-Length: 116
>>> Contact: <sip:192.168.68.12>
>>> Call-ID: F1C6B16E-51A7-4987-A2B9-128C2302378C at 192.168.68.12
>>> Content-Type: application/sdp
>>> Max-Forwards: 70
>>> From: "Ranga Rao Vutukuru"<sip:sridhar at 192.168.68.15>;tag=21632105
>>> CSeq: 2 INVITE
>>> To: <sip:ranga at 192.168.68.6>
>>> Via: SIP/2.0/UDP 192.168.68.12:5060
>>> Proxy-Authorization: Digest
>>> username="sridhar",realm="asterisk",nonce="25230b01",uri="sip:ranga at 192.168. 
>>>
>>> 68.6",response="bb1576d7abea9f08c07d598c7d6686a0"
>>>
>>> v=0
>>> o=- 3279257833 3279257833 IN IP4 192.168.68.12
>>> s=-
>>> c=IN IP4 192.168.68.12
>>> t=0 0
>>> m=audio 16390 RTP/AVP 8 0
>>>
>>> 11 headers, 6 lines
>>> Using latest request as basis request
>>> Sending to 192.168.68.12 : 5060 (non-NAT)
>>> Found audio format ALAW
>>> Found audio format UNKN
>>> Capabilities: us - 524302, them - 12/0, combined - 12
>>> Non-codec capabilities: us - 1, them - 0, combined - 0
>>> Looking for ranga in pandora
>>> list_route: hop: <sip:192.168.68.12>
>>> Transmitting (no NAT):
>>> SIP/2.0 100 Trying
>>> Via: SIP/2.0/UDP 192.168.68.12:5060
>>> From: "Ranga Rao Vutukuru"<sip:sridhar at 192.168.68.15>;tag=21632105
>>> To: <sip:ranga at 192.168.68.6>;tag=as62db81f5
>>> Call-ID: F1C6B16E-51A7-4987-A2B9-128C2302378C at 192.168.68.12
>>> CSeq: 2 INVITE
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>>> Contact: <sip:ranga at 192.168.68.15>
>>> Content-Length: 0
>>>
>>>
>>>  to 192.168.68.12:5060
>>>     -- Executing SetGlobalVar("SIP/sridhar-51cd", "sipto=ranga") in new
>>> stack
>>>     -- Setting global variable 'sipto' to 'ranga'
>>>     -- Executing SetGlobalVar("SIP/sridhar-51cd", "sipdom=") in new 
>>> stack
>>>     -- Setting global variable 'sipdom' to ''
>>>     -- Executing GotoIf("SIP/sridhar-51cd", "0?30|1:5|1") in new stack
>>>     -- Goto (pandora,5,1)
>>>     -- Executing GotoIf("SIP/sridhar-51cd", "0?20|1:10|1") in new stack
>>>     -- Goto (pandora,10,1)
>>>     -- Executing Dial("SIP/sridhar-51cd", "SIP/ranga@") in new stack
>>>   == Everyone is busy at this time
>>>     -- Executing Hangup("SIP/sridhar-51cd", "") in new stack
>>>   == Spawn extension (pandora, 10, 2) exited non-zero on 
>>> 'SIP/sridhar-51cd'
>>>     -- Executing SetGlobalVar("SIP/sridhar-51cd", "sipto=h") in new 
>>> stack
>>>     -- Setting global variable 'sipto' to 'h'
>>>     -- Executing SetGlobalVar("SIP/sridhar-51cd", "sipdom=") in new 
>>> stack
>>>     -- Setting global variable 'sipdom' to ''
>>>     -- Executing GotoIf("SIP/sridhar-51cd", "1?30|1:5|1") in new stack
>>>     -- Goto (pandora,30,1)
>>>     -- Executing Hangup("SIP/sridhar-51cd", "") in new stack
>>>   == Spawn extension (pandora, 30, 1) exited non-zero on 
>>> 'SIP/sridhar-51cd'
>>> Reliably Transmitting (no NAT):
>>> SIP/2.0 403 Forbidden
>>> Via: SIP/2.0/UDP 192.168.68.12:5060
>>> From: "Ranga Rao Vutukuru"<sip:sridhar at 192.168.68.15>;tag=21632105
>>> To: <sip:ranga at 192.168.68.6>;tag=as62db81f5
>>> Call-ID: F1C6B16E-51A7-4987-A2B9-128C2302378C at 192.168.68.12
>>> CSeq: 2 INVITE
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
>>> Contact: <sip:ranga at 192.168.68.15>
>>> Content-Length: 0
>>>
>>>
>>>  to 192.168.68.12:5060
>>> Sip read: CLI>
>>> ACK sip:ranga at 192.168.68.6 SIP/2.0
>>> Content-Length: 0
>>> Call-ID: F1C6B16E-51A7-4987-A2B9-128C2302378C at 192.168.68.12
>>> CSeq: 2 ACK
>>> From: "Ranga Rao Vutukuru"<sip:sridhar at 192.168.68.15>;tag=21632105
>>> To: <sip:ranga at 192.168.68.6>;tag=as62db81f5
>>> Via: SIP/2.0/UDP 192.168.68.12:5060
>>>
>>>
>>> 7 headers, 0 lines
>>> localhost*CLI>
>>>
>>> ----- Original Message -----
>>> From: "Olle E. Johansson" <oej at edvina.net>
>>> To: <asterisk-users at lists.digium.com>
>>> Sent: Monday, December 01, 2003 2:16 PM
>>> Subject: Re: [Asterisk-Users] Asterisk as SIP Proxy
>>>
>>>
>>>> ranga wrote:
>>>>
>>>>> This is the complete extensions.conf. I wasnt getting the SIPDOMAIN
>>>
>>>
>>> right.
>>>
>>>>> Rest of your script/configuration works only if ${SIPDOMAIN} works
>>>>> Am I missing anything in this? I had the latest CVS checkout this
>>>
>>>
>>> morning,
>>>
>>>>> i.e., 1st Dec. 12.00 Noon GMT +5.30.
>>>>
>>>>
>>>> Ranga,
>>>> I agree, seems like the client is not sending an INVITE that Asterisk
>>>> is able to parse the SIPDOMAIN from.
>>>>
>>>> Turn on SIP DEBUG in your Asterisk CLI and catch the INVITE from the
>>>
>>>
>>> client.
>>>
>>>> Check if the invite goes to user at domain or only to "user" without a
>>>
>>>
>>> domain?
>>>
>>>> I haven't got sjphone, so I can't try myself.
>>>>
>>>> Please add a SIP DEBUG output with the INVITE.
>>>>
>>
>>
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> 
> 
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-- 
*** Olle E. Johansson, oej at edvina.net

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