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:[EMAIL PROTECTED] SIP/2.0 Content-Length: 116 Contact: <sip:192.168.68.12> Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp Max-Forwards: 70 From: "Ranga Rao Vutukuru"<sip:[EMAIL PROTECTED]>;tag=21632105 CSeq: 1 INVITE To: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=21632105 To: <sip:[EMAIL PROTECTED]>;tag=as78933dd8 Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED] SIP/2.0 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK From: "Ranga Rao Vutukuru"<sip:[EMAIL PROTECTED]>;tag=21632105 To: <sip:[EMAIL PROTECTED]>;tag=as78933dd8 Via: SIP/2.0/UDP 192.168.68.12:5060
7 headers, 0 lines Sip read: CLI> INVITE sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 116 Contact: <sip:192.168.68.12> Call-ID: [EMAIL PROTECTED] Content-Type: application/sdp Max-Forwards: 70 From: "Ranga Rao Vutukuru"<sip:[EMAIL PROTECTED]>;tag=21632105 CSeq: 2 INVITE To: <sip:[EMAIL PROTECTED]> Via: SIP/2.0/UDP 192.168.68.12:5060 Proxy-Authorization: Digest username="sridhar",realm="asterisk",nonce="25230b01",uri="sip:[EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=21632105 To: <sip:[EMAIL PROTECTED]>;tag=as62db81f5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=21632105 To: <sip:[EMAIL PROTECTED]>;tag=as62db81f5 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0
to 192.168.68.12:5060 Sip read: CLI> ACK sip:[EMAIL PROTECTED] SIP/2.0 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 2 ACK From: "Ranga Rao Vutukuru"<sip:[EMAIL PROTECTED]>;tag=21632105 To: <sip:[EMAIL PROTECTED]>;tag=as62db81f5 Via: SIP/2.0/UDP 192.168.68.12:5060
7 headers, 0 lines localhost*CLI>
----- Original Message ----- From: "Olle E. Johansson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> 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 [EMAIL PROTECTED] 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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