search for: 9_outside

...xten => 9002,1,Macro(exten-vm,9002@default,9002) exten => ${VM_PREFIX}9002,1,Macro(vm,9002) exten => abhijit,1,Macro(exten-vm,abhijit@default,abhijit) exten => ${VM_PREFIX}abhijit,1,Macro(vm,abhijit) [outbound-allroutes] include => outbound-allroutes-custom include => outrt-001-9_outside include => outrt-002-outgoingFWD [outbound-trunks] include => outbound-trunks-custom exten => _${DIAL_OUT_1}.,1,Macro(dialout,1,${EXTEN}) [outrt-001-9_outside] include => outrt-001-9_outside-custom exten => _9.,1,Macro(dialout-trunk,1,${EXTEN:1}) exten => _9.,2,Macro(outisbusy)...

Dear All, How can i add this extentions eg: 145,146,147,201,202 to allow dialout call, i've been add this ext to GROUP variable like this GROUP = 145,146,147,201,202 [outrt-001-9_outside] include => outrt-001-9_outside-custom exten => _9.,1,GotoIf($[${CALLERIDNUM} != ${GROUP} } ]?105) ;Exceeded? exten => _9.,2,Macro(dialout-trunk,1,${EXTEN:1}) exten => _9.,3,Macro(outisbusy) ; No available circuits exten => _9.,105,Hangup but only ext 145 can dial, the others w...

I want to say thanks to everyone for the help so far. I figured out a way to modify some AAH code that worked for me (well sort of). The line I modified is s,14 in macro-dialout-trunk. Then I just added a variable and passed it from 9_outside. I just have one last problem. This waits for an answer not ringing. So if the called party has a long ring to voice mail the call is dropped and goes out the PRI. Does anyone know of a way to listen for ringing on an IAX2 channel? [9_outside] exten => _9Z.,1,Macro(dialout-trunk,"tr...

...ck to the Digium FXO port, I cannot dial 9 and the phone number to route my call to the PSTN through the legacy PBX. Looking at the AMP (Asterisk Management Portal)=>Outbound Route, I have two routes created: PBX=>to several legacy PBX extensions: 250, 270, 280 (these are the dial patterns) 9_outside=>to the default dial pattern included with AAH: 9|. (that is the sole dial pattern) I wonder if the digits get dialed too fast to connect to the PSTN? Can I put a pause in somehow? I can see that Asterisk does grab the outside line. When I dial, I get the following message, which I think is...

...n=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no faxdetect=both faxdetect=incoming faxdetect=outgoing faxdetect=no context=default ; Points to the default context of your extensions.conf channel => 1-15,17-31,32-46,48-62; for E1 i've configured the outgoing calls [outrt-001-9_outside] include => outrt-001-9_outside-custom exten => _9XXXXXXXXXX,1,Macro(dialout-trunk,1,${EXTEN:1},) exten => _9XXXXXXXXXX,2,Macro(outisbusy) ; No available circuits if i try to call i get: Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Setting NAT on RTP to 0 Feb 13 06:19:44 DEBUG[3637] chan_sip....

...se they are answering and not sending a error code. If you are using AAH code then this waits 10 seconds on your Voip then times out and goes to PSTN. You can modify for your needs The pertinent line is 14 in macro-dialout-trunk I am going to clean it up and repost under my original question [9_outside] exten => _9Z.,1,Macro(dialout-trunk,"trunknumerhere",${EXTEN:1},,10) exten => _9Z.,2,Macro(dialout-trunk, "trunknumerhere",${,${EXTEN:1},) exten => _9Z.,3,Macro(outisbusy) ; No available circuits [macro-dialout-trunk] exten => s,1,GotoIf($[foo${ARG3} = foo]?3:2))...

...n=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no faxdetect=both faxdetect=incoming faxdetect=outgoing faxdetect=no context=default ; Points to the default context of your extensions.conf channel => 1-15,17-31,32-46,48-62; for E1 i've configured the outgoing calls [outrt-001-9_outside] include => outrt-001-9_outside-custom exten => _9XXXXXXXXXX,1,Macro(dialout-trunk,1,${EXTEN:1},) exten => _9XXXXXXXXXX,2,Macro(outisbusy) ; No available circuits if i try to call i get: Feb 13 06:19:44 DEBUG[3637] chan_sip.c: Setting NAT on RTP to 0 Feb 13 06:19:44 DEBUG[3637] chan_sip....

...uthentication is sent to my SIP Provider, but how do I integrate this in my call. Above all, I have found several articles on the internet stating this WARNING[1563], but they all have more information after the INVITE than I do. Below you can find part of my extensions.conf file: [outrt-001-9_outside] exten => _XXXXXXXXXX,1,SetCallerID(31437110323) exten => _XXXXXXXXXX,2,SetCIDName(31437110323) exten => _XXXXXXXXXX,3,SetCIDNum(31437110323) exten => _XXXXXXXXXX,4,Dial(SIP/0${EXTEN:1}@budgetphone.nl) ;exten => _XXXXXXXXXX,5,Playback(invalid) exten => _XXXXXXXXXX,5,Hangup...

I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it. Can I make this thing into MGCP gateway or even a SIP gateway for asterisk? Seems like it should bee useful for something! I'm perfectly happy to do my homework, but also don't feel thee need to reinvent the wheel! So, links with relevant info would be appreciated. If there is a config for a 2621 being used as a gateway

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 this message give me when I calling a number than actually not busy: "Dial failed due to trunk reporting BUSY - giving up" max channel is unlimited and sometimes it dial number ok but most of the time it gives me this error. Please inform me how can solve this problem. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I am trying to connect my asterisk box with freepbx to PSTN. I have purchased x100p FXO card and installed in my asterisk server. My freepbx detected the x100p FXO card and i can see the card specific details in command line. I have configured the following things. 1. OUTBOUND caller id and Dialing rules in Freepbx. 2. INBOUND route When i call to the PSTN number before