[Asterisk-Users] Asterisk as SIP Proxy

[John Todd]

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.

JT


>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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