[Asterisk-Users] Asterisk as SIP Proxy

Mon Dec 1 07:51:42 MST 2003

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.
>>