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...Declare the file path before you record it. $path = "/usr/local/apache/htdocs/demo/sound/myapp.$date.wav"; $AGI->exec('Record',$path:wav"); -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of AstGrp Sent: Sunday, May 09, 2004 3:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AGI Assitance 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('...

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 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 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)

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 =>

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 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

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 -

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

When we receive or make a call to the outside - they can hear us, but we cant hear them. It may work 1 of 20 times. I have set canreinvite=no and looked at many sites but cannot track down this problem. Current setup: Isdn Eicon Diva card / Capi -> Asterisk --> network. I have tried adjusting the RTP port in rtp.conf with the Cisco default ports, no luck. Anyone had this

"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas

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