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I have seen this posted around the list a few times, but have never found my exact error or even a solution to this. I was getting this error every few seconds...... After moving the card to a different PCI slot and turning off some devices that are not needed. I was able to get the messages down to about every 10 minutes and the T1 card now is sitting in it's own intterupt. Apr 25 23:49:15

I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to externip = 69.132.68.17 ; Address

Still getting the same error. Apr 29 11:57:49 WARNING[1125329600]: chan_sip.c:503 retrans_pkt: Maximum retries exceeded on call 6b8b4567327b23c6643c986966334873@211.28.255.135 for seqno 102 (Critical Request) please advise anyone!!!!!someone!!! jai

I am trying to figure this out... I'm sure it's simple, but I can't think of it right now.... In my AGI Script I am doing this... (This is done in Perl) $AGI->exec('Record', "/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav"); And after this is done.. I want to get the name of the file it created so I can store it in a database. Any thoughts.... Thanks, -gcc

I was wondering if this is possible. I have a situation where I am connecting to a third party voicemail system from asterisk. I know this does not make since to everyone, but it has to be this way. Basically - I have an application that runs on the Asterisk system and when an employee calls into this system, they have an option to check there voicemail. This is where it needs to go over to

I have a need to interface Asterisk with a VoiceWorks voicemail system. I was wondering what kind of card would be needed either a FXO or FXS interface? Any help would be appreciated. Thanks, -gcc

I am trying to implement a rollover of extensions. exten => 3000,1,GotoIf($[${line1} = Congestion]?3:2) exten => 3000,2,Dial(${line1},15,rt) exten => 3000,3,GotoIf($[${line2} = Congestion]?5:4) exten => 3000,4,Dial(${line2},15,rt) exten => 3000,5,GotoIf($[${line3} = Congestion]?7:6) exten => 3000,6,Dial(${line3},15,rt) exten => 3000,7,GotoIf($[${line4} = Congestion]?1:8)

Can you post some of your sip configs and your extension configs. Thanks, -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ryan Parlee Posted At: Sunday, April 04, 2004 12:10 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT Subject: [Asterisk-Users] Asterisk -

Let me lay it out for you.... Call comes in over a T1 - Signal is em_w. The extension is seen as *<callerid>*<last 4 digits of number being called>*. Which is fine in it self. I have my extension.conf file set up as follows... [did] ; Receive call as *<calling>*<called> exten => _.,1,Answer exten => _.,2,Cut(CALLING=EXTEN,*,2) exten =>

That is not working... I tried like you mentioned it and even a few different ways and will not create the file at all.... Am I doing something wrong... -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Todd Lieberman Posted At: Sunday, May 09, 2004 4:00 PM Posted To: Asterisk User Group Conversation: