search for: the_num

...oto(s-${DIALSTATUS},1) ; This is a custom trunk. Substitute $OUTNUM$ with the actual number and rebuild the dialstring ; example trunks: "AMP:CAPI/XXXXXXXX:b$OUTNUM$,30,r", "AMP:OH323/$OUTNUM$@XX.XX.XX.XX:XXXX" exten => s,19,Cut(pre_num=OUT_${ARG1},$,1) exten => s,20,Cut(the_num=OUT_${ARG1},$,2) ; this is where we expect to find string OUTNUM exten => s,21,Cut(post_num=OUT_${ARG1},$,3) exten => s,22,GotoIf($[${the_num} = OUTNUM]?23:24) ; if we didn't find "OUTNUM", then skip to Dial exten => s,23,SetVar(the_num=${OUTNUM}) ; replace "OUTNUM&quot...

...uot;SIP/16000-00000002", > "pre_num=AMP:h323/Avaya/") in new stack > [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set > [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing > [s at macro-dialout-trunk:22] Set("SIP/16000-00000002", "the_num=OUTNUM") in > new stack > [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set > [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing > [s at macro-dialout-trunk:23] Set("SIP/16000-00000002", "post_num=@ > 10.100.7.15:1720") in new stack...

I currently have: Telco-PRI ---- Panasonic DBS576 PBX ---- E&M wink T1 ---- Asterisk. I have configured the Panasonic to forward my Asterisk DIDs to the Asterisk extensions over the T1. I do not get DID nor CID on the Asterisk, so I want to use PRI between the PBXs. I do not want to pay for another PRI card for the Panasonic. (T1 and PRI are different cards) I see this as my least

...Goto(s-${DIALSTATUS},1) ; This is a custom trunk. Substitute $OUTNUM$ with the actual number and rebuild the dialstring ; example trunks: "AMP:CAPI/XXXXXXXX:b$OUTNUM$,30,r", "AMP:OH323/$OUTNUM$@XX.XX.XX.XX:XXXX" exten => s,19,Cut(pre_num=OUT_${ARG1},$,1) exten => s,20,Cut(the_num=OUT_${ARG1},$,2) ; this is where we expect to find string OUTNUM exten => s,21,Cut(post_num=OUT_${ARG1},$,3) exten => s,22,GotoIf($[${the_num} = OUTNUM]?23:24) ; if we didn't find "OUTNUM", then skip to Dial exten => s,23,SetVar(the_num=${OUTNUM}) ; replace "OUTNUM"...

...) OK. when I try to access an external line, asterisk crashes with a segmentation fault; the dial string is correct ... -- Executing GotoIf("SIP/567-bb09", "1?20:21") in new stack -- Goto (macro-dialout-trunk,s,20) -- Executing SetVar("SIP/567-bb09", "the_num=3481303063") in new stack -- Executing Dial("SIP/567-bb09", "misdn/1/3481303063") in new stack -- Called 1/3481303063 Ouch ... error while writing audio data: : Broken pipe Segmentation fault (core dumped) I am using Suse Linux 10, and I switched to default kernel...

Hi, How could I possibly inform incoming callers that the number they'd dialed is monitored and recorded. I wanted that when a call-in or call-out is made, a playback will be played to inform caller & callee that thier line is monitored prior to start conversation. Thanks. Angel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm