Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... sip.conf extract : [gw001] type=friend host=dynamic defaultip=192.168.0.12 nat=no dtmfmode=rfc2833 canreinvite=yes qualify=no context=tlsgw extensions.conf extract (from the contact [tlsgw]) : exten => 57228047,Dial(SIP/cs001,40,tr) ... Sip read: INVITE sip:57228047@192.168.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 <sip:478758923@192.168.0.12>;tag=0000-0002-3A81BAD9 To: <sip:57228047@192.168.0.15> Max-Forwards: 70 Call-ID: 0001-0002-94F67DB7-0@192.168.0.12 CSeq: 83606 INVITE Contact: <sip:478758923@192.168.0.12:5060;maddr=192.168.0.12> Supported: replaces Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER Accept-Language: en Content-Type: application/sdp Remote-Party-ID: 478758923 <sip:478758923@192.168.0.12>;party=calling;screen=no;privacy=off Content-Length: 178 v=0 o=Vega50 3 1 IN IP4 192.168.0.12 s=Sip Call t=0 0 m=audio 10004 RTP/AVP 8 0 18 c=IN IP4 192.168.0.12 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 14 headers, 9 lines Using latest request as basis request Sending to 192.168.0.12 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMA Found description format PCMU Found description format G729 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 <sip:478758923@192.168.0.12>;tag=0000-0002-3A81BAD9 To: <sip:57228047@192.168.0.15>;tag=as1fa83a23 Call-ID: 0001-0002-94F67DB7-0@192.168.0.12 CSeq: 83606 INVITE ser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:57228047@192.168.0.15> Content-Length: 0 to 192.168.0.12:5060 dkmapbx*CLI> Sip read: ACK sip:57228047@192.168.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 <sip:478758923@192.168.0.12>;tag=0000-0002-3A81BAD9 To: <sip:57228047@192.168.0.15> Max-Forwards: 70 Call-ID: 0001-0002-94F67DB7-0@192.168.0.12 CSeq: 83606 ACK Contact: <sip:478758923@192.168.0.12:5060;maddr=192.168.0.12> Content-Length: 0 9 headers, 0 lines DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer.
Michael Devenijn wrote:> Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... > > > sip.conf extract : > > [gw001] > type=friend > host=dynamic > defaultip=192.168.0.12 > nat=no > dtmfmode=rfc2833 > canreinvite=yes > qualify=no > context=tlsgw > > > > extensions.conf extract (from the contact [tlsgw]) : > > exten => 57228047,Dial(SIP/cs001,40,tr) > ...Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Asterisk isn't looking in the context tlsgw, for some reason it checks in the "sip" context. If this is your default context, Asterisk doesn't connect the incoming call with gw001. You have host=dynamic - is the gateway registred with Asterisk at all? /O
sorry, the sip extract is from a previous test now i get the same problem but with looking for 57228047 in tlsgw and it's the same error, it searching in this direction : why are the 2 ast values 0 ?? Non-codec capabilities: us - 1, them - 0, combined - 0 -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Olle E. Johansson Sent: Saturday, March 20, 2004 11:05 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with Vegastream 50 BRI Michael Devenijn wrote:> Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... > > > sip.conf extract : > > [gw001] > type=friend > host=dynamic > defaultip=192.168.0.12 > nat=no > dtmfmode=rfc2833 > canreinvite=yes > qualify=no > context=tlsgw > > > > extensions.conf extract (from the contact [tlsgw]) : > > exten => 57228047,Dial(SIP/cs001,40,tr) > ...Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Asterisk isn't looking in the context tlsgw, for some reason it checks in the "sip" context. If this is your default context, Asterisk doesn't connect the incoming call with gw001. You have host=dynamic - is the gateway registred with Asterisk at all? /O _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer.
This was posted to me by Vegastream tech support in regards to your earlier question (I emailed them your question last week), sorry I'm just getting familiar with both boxes so I'm not able to help you at this stage I have just signed a deal for distribution of the vegastream here in Australia and for anyone here on the list who is interested it will be shortly certified with Comindico for use on the new ecall network. Cheers, Dean Hi, We don't have any particular hands-on experience with Asterisk ourselves, although a number of customers use it. My main observation would be: - The asterisk server isn't finding the phone number 57228047 and is returning a 404 (obviously) - As the follow-up poster observed, asterisk has "tlsgw" set as the default context and has entered the phone number extension for tlsconf, but is using the "sip" context instead, hence doesn't find the number - Quite possibly this is because the Vega doesn't do TLS (Transport Layer Security), and therefore that context isn't applicable, though that's just a guess since I haven't used asterisk... Cheers, Bryan -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Michael Devenijn Sent: Sunday, 21 March 2004 2:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Problem with Vegastream 50 BRI Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ... sip.conf extract : [gw001] type=friend host=dynamic defaultip=192.168.0.12 nat=no dtmfmode=rfc2833 canreinvite=yes qualify=no context=tlsgw extensions.conf extract (from the contact [tlsgw]) : exten => 57228047,Dial(SIP/cs001,40,tr) ... Sip read: INVITE sip:57228047@192.168.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 <sip:478758923@192.168.0.12>;tag=0000-0002-3A81BAD9 To: <sip:57228047@192.168.0.15> Max-Forwards: 70 Call-ID: 0001-0002-94F67DB7-0@192.168.0.12 CSeq: 83606 INVITE Contact: <sip:478758923@192.168.0.12:5060;maddr=192.168.0.12> Supported: replaces Allow: INVITE,ACK,BYE,CANCEL,INFO,NOTIFY,OPTIONS,REFER Accept-Language: en Content-Type: application/sdp Remote-Party-ID: 478758923 <sip:478758923@192.168.0.12>;party=calling;screen=no;privacy=off Content-Length: 178 v=0 o=Vega50 3 1 IN IP4 192.168.0.12 s=Sip Call t=0 0 m=audio 10004 RTP/AVP 8 0 18 c=IN IP4 192.168.0.12 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 14 headers, 9 lines Using latest request as basis request Sending to 192.168.0.12 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Found audio format UNKN Found description format PCMA Found description format PCMU Found description format G729 Capabilities: us - 524302, them - 268/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 57228047 in sip Transmitting (no NAT): SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 <sip:478758923@192.168.0.12>;tag=0000-0002-3A81BAD9 To: <sip:57228047@192.168.0.15>;tag=as1fa83a23 Call-ID: 0001-0002-94F67DB7-0@192.168.0.12 CSeq: 83606 INVITE ser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:57228047@192.168.0.15> Content-Length: 0 to 192.168.0.12:5060 dkmapbx*CLI> Sip read: ACK sip:57228047@192.168.0.15:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.12:5060;branch=z9hG4bK-vega1-000A-0001-0002-1A36AF7D From: 478758923 <sip:478758923@192.168.0.12>;tag=0000-0002-3A81BAD9 To: <sip:57228047@192.168.0.15> Max-Forwards: 70 Call-ID: 0001-0002-94F67DB7-0@192.168.0.12 CSeq: 83606 ACK Contact: <sip:478758923@192.168.0.12:5060;maddr=192.168.0.12> Content-Length: 0 9 headers, 0 lines DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Armand A. Verstappen
2004-Mar-22 11:35 UTC
[Asterisk-Users] Problem with Vegastream 50 BRI
On Sat, 2004-03-20 at 16:36, Michael Devenijn wrote:> Here is a sip log from my vegastream 50BRI to my asterisk box and i can't figure out why the call doesn't go trough ...> extensions.conf extract (from the contact [tlsgw]) : > > exten => 57228047,Dial(SIP/cs001,40,tr)The above line does not look like a valid extension line, the priority is missing. That _could_ prevent the context tlsgw from being loaded, which in turn might cause your installation to fallback to the default context. You may want to inspect the output of a 'show dialplan' to see if your tlswg context is loaded or not. Apart from that, you may want to increase logging in /etc/asterisk/logger.conf for a default installation: debug => debug console => notice,warning,error,debug messages => notice,warning,error This will cause debug messages to be show on the console where you are running asterisk, and to log them to a file /var/log/asterisk/debug, while notice, warning and errors will be logged to /var/log/asterisk/messages. These files will be your friends in debugging. wkr, -- Envida http://www.envida.net/ Armand A. Verstappen Vleutenseweg 86 armand@nl.envida.net 3532 HM Utrecht tel: +31 (0)30 299 2109 The Netherlands fax: +31 (0)30 299 2108 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040322/c7bdb743/attachment.pgp