search for: outnum

I'm having trouble getting outbound calls going with aah 1.3 and a tdm400p w/ 4 FXO. Incoming calls work fine, outbound I get this: -- Executing SetVar("SIP/231-af2b", "OUTNUM=6643955") in new stack -- Executing Cut("SIP/231-af2b", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/231-af2b", "0?19") in new stack -- Executing Dial("SIP/231-af2b", "ZAP/g0/6643955") in new stack -- Call...

Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards "standardizing" installed modules, functionalities, tools etc. The "wall" we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to

...o add extensions and > trunks manually. > I apologize for spamming the list... I failed to mention that Custom Trunks support exists in the current version of AMP (1.10.008). Here is the text from the "Custom Dial String" tooltip: Define the custom Dial String. Include the token $OUTNUM$ wherever the number to dial should go. examples: CAPI/XXXXXXXX:b$OUTNUM$,30,r H323/$OUTNUM$@XX.XX.XX.XX OH323/$OUTNUM$@XX.XX.XX.XX:XXXX vpb/1-1/$OUTNUM$ -- Jason Becker Director & CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca _____________...

I installed oh323 and everything seemed to go smoothly (compile & everything upto calling through using oh323). I must admit, there is some behavior that's doesn't seem right but generally, I'm able to dial-out of any oh323 device whether to an extension or to a trunk. Audio is sometimes muted when dialing out until the extension or dialed number answers. Sound quality is good

...p(${OUTMAXCHANS_${ARG1}}) ; if we've used up the max channels, continue at (n+101) exten => s,8,SetVar(DIAL_NUMBER=${ARG2}) exten => s,9,SetVar(DIAL_TRUNK=${ARG1}) exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial string for this trunk exten => s,11,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ; OUTNUM is the final dial number exten => s,12,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are prefixed with "AMP:" exten => s,13,GotoIf($[${custom} = AMP]?16) ;exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial exten => s...

...p(${OUTMAXCHANS_${ARG1}}) ; if we've used up the max channels, continue at (n+101) exten => s,8,SetVar(DIAL_NUMBER=${ARG2}) exten => s,9,SetVar(DIAL_TRUNK=${ARG1}) exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial string for this trunk exten => s,11,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ; OUTNUM is the final dial number exten => s,12,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are prefixed with "AMP:" exten => s,13,GotoIf($[${custom} = AMP]?16) ;exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial exten => s...

...UTMAXCHANS_${ARG1}}) ; if we've used up the max channels, continue at 110 (n+101) exten => s,8,SetVar(DIAL_NUMBER=${ARG2}) exten => s,9,SetVar(DIAL_TRUNK=${ARG1}) exten => s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial string for this trunk exten => s,11,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ; OUTNUM is the final dial number exten => s,12,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are prefixed with "AMP:" exten => s,13,GotoIf($[${custom} = AMP]?16) exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial exten =>...

...TMAXCHANS_${ARG1}}) ; if we've used up the max channels, continue at 109 (n+101) exten => s,11,SetVar(DIAL_NUMBER=${ARG2}) exten => s,12,SetVar(DIAL_TRUNK=${ARG1}) exten => s,13,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper dial string for this trunk exten => s,14,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ; OUTNUM is the final dial number exten => s,15,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are prefixed with "AMP:" exten => s,16,GotoIf($[${custom} = AMP]?19) exten => s,17,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial exten => s,18,...

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

...uot;SIP/16000-00000002", > "0|AGI|fixlocalprefix") in new stack > [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: ExecIf > [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing > [s at macro-dialout-trunk:13] Set("SIP/16000-00000002", "OUTNUM=18151") 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:14] Set("SIP/16000-00000002", "custom=AMP") in new > stack > [Jun 16 15:2...

...s. Number of My date value are about 4000, It is continuous time series, 3 days, the interveal is 5 minutes, (2002-09-22 12:35:00 pm to 2002-09-22 3:40:00 pm) How can i plot it in R? I tried the regular way: formattime <- strptime(time, format= "%d-%b-%y %I:%M %p") plot(formattime, outNum) However the x axes doen't give me the hours by each three days, instead it is "mon" "tue" etc. What should I do? Thank you! --------------------------------- [[alternative HTML version deleted]]

Hi All, in regards to my previous queries about call transfers not working from inside, several days of searching turned up this posting: I got this to work by editing the line exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) to say exten => s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt) in extensions.conf seems like many people have had this issue in the past, I guess it's AMP related, as I added those options in the 'general settings' dialog in AMP, but it would never work when the call originated inte...

Hi I have asterisk 1.4 I want to make a MGCP trunk as a client to connect to a provider who is using MGCP protocol, he provided me with user & password, I tried a custom trunk: MGCP/$OUTNUM$@user:password at 66.152.163.106:4000 Not seems to help, Any suggestions plz?

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

Hi I've got a dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out using the acopy2 test utility. I'm having trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to look ? I can attach conf files etc. if needed. Asterisk says it has 30 capi channels available,

..., 4) exited non-zero on 'Zap/14-1' in macro 'hangupcall' == Spawn extension (default, s, 2) exited non-zero on 'Zap/14-1' -- Hungup 'Zap/14-1' ---------------------------- ------------------outbound call -- Executing SetVar("SIP/4901-cd04", "OUTNUM=mynum") in new stack -- Executing Cut("SIP/4901-cd04", "custom=OUT_1|:|1") in new stack -- Executing GotoIf("SIP/4901-cd04", "0?19") in new stack -- Executing Dial("SIP/4901-cd04", "ZAP/g0/mynum") in new stack -- Calle...

...AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/146-b78060b0", "OUTNUM=6536595") in new stack -- Executing Set("SIP/146-b78060b0", "custom=ZAP/g0") in new stack -- Executing GotoIf("SIP/146-b78060b0", "0?16") in new stack -- Executing Dial("SIP/146-b78060b0", "ZAP/g0/6536595|120|tr") in new s...

OK, im pulling my hair out, everything looks configured right, deleted, and started over, etc, etc. but can't seem to get this to work Gtalk.conf [general] context=google-in allowguest=yes bindaddr=192.168.xxx.xxx extenip=96.254.xxx.xxx [guest] context=google-in disallow=all allow=ulaw allow=g729 connection=jp_jabber jabber.conf [general] debug=yes

...uot;) in new stack -- Executing DeadAGI("SIP/200-09fc1698", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/200-09fc1698", "OUTNUM=637574972") in new stack -- Executing Set("SIP/200-09fc1698", "custom=ZAP/1") in new stack -- Executing GotoIf("SIP/200-09fc1698", "0?customtrunk") in new stack -- Executing Dial("SIP/200-09fc1698", "ZAP/1/637574972|300|r&qu...

...uot;DIAL_TRUNK=4") in new stack -- Executing AGI("IAX2/4414-6", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("IAX2/4414-6", "OUTNUM=xxxnnnnnnn") in new stack -- Executing Set("IAX2/4414-6", "custom=ZAP/g0") in new stack -- Executing GotoIf("IAX2/4414-6", "0?16") in new stack -- Executing Dial("IAX2/4414-6", "ZAP/g0/xxxnnnnnnn|120|r") in new stack ==...