[Asterisk-Users] Asterisk as SIP Proxy

[ranga]

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.
>
> /Olle
>
> _______________________________________________
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> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>