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An application I am running provides a dial tone to my users, read 9 digits, checks whether or not the called party number should be allowed and then dials out using overlap dialing on a pri channel. I.e. exten => _XXXXXXXXX,1,AGI(pm-check-destination.agi) exten => _XXXXXXXXX,2,Dial,Zap/g1/BYEXTENSION|60|CH The AGI-Skript takes about 0.3 to 0.5 seconds (it does a number of rather complex database queries). Any digits dialed during this time are lost. I tried to retrieve those digits at the end of the AGI-Skript b...

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

Is there any way to receive and send faxes using a T100 card? If so how is it done? Gene Kochanowsky Solution Sciences, Inc.

Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/3ff8a388/attachment.htm

...conf" and "h323.conf" files are: *************************************** extensions.conf >>>> *************************************** [default] ..... ..... [outgoing] exten=>_7XX,1,Goto(voip-h323|${EXTEN}|1) exten=>_*XX,1,Goto(servicios|${EXTEN}|1) exten=>_XXXXXXXXX,1,Dial(Zap/1/${EXTEN}|30) exten=>_XXXXXXXXX,2,Playback(invalid) exten=>_XXXXXXXXX,3,Hungup() exten=>_X,1,Playback(invalid) exten=>_X,2,Hungup exten=>_XX,1,Playback(invalid) exten=>_XX,2,Hungup exten=>_XXXX,1,Playback(invalid) exten=>_XXXX,2,Hungup exten=>_XXXXX,1,Playbac...

...CST) > > let's say you have one T1 span configured like this > in zapata.conf > > context=incoming > group = 1 > channel => 1-23 > > then in extensions.conf > [incoming] > exten => fax,1,Dial,Zap/25 #FXS port that fax is > plugged to > exten => _XXXXXXXXX,1,... (the rest) > > when asterisk detects fax tones on incoming call > it's going > to look for fax extension in the channel context. If > it finds > it then the fax call is going to be routed according > to > the fax extension rule. > > regards > Martin >...

...XXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _0XXXXXXXXX,2,MixMonitor(/var/spool/asterisk/astrec/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten => _0XXXXXXXXX,3,Dial(${TRUNK}/${EXTEN},,tTo) exten => _0XXXXXXXXX,4,Hangup exten => _XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _XXXXXXXXX,2,MixMonitor(/var/spool/asterisk/astrec/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten => _XXXXXXXXX,3,Dial(${TRUNK}/${EXTEN},,tTo) exten => _XXXXXXXXX,4,Hangup Console is full duple...

Friends, Last week, Hans Petter Selansky alerted us of a potential security issue in all releases of Asterisk. In fact, it doesn't involve the code, but the most common way to construct dialplans. If you have something like this in your Asterisk, you need to update your dialplans: [incoming-from-voip] exten => _X., 1, dial(SIP/${EXTEN}) Many VoIP protocols support a large character set,

...=ilbc host=217.221.182.66 extenxion.conf: [general] static=yes writeprotect=yes [globals] [macro-stdiax] exten => s,1,Dial(IAX2/${ARG1}|20|Ttr) [fullaccess] include => local [local] exten => 101,1,Macro(stdiax,first,${EXTEN}) exten => 102,1,Macro(stdiax,second,${EXTEN}) exten => _XXXXXXXXXX,1,Dial(ZAP/1/${EXTEN}) exten => _XXXXXXXXX,1,Dial(ZAP/1/${EXTEN}) exten => _XXX,1,Dial(ZAP/1/${EXTEN}) exten => _011.,1,Dial(IAX2/megavista/${EXTEN:3}) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachme...

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

Could you kick me, I can't dial more then 9 digits. Is anyone some default length of extensions or dialed number. Thanks, Bob.