search for: outrt

...vm,9001) exten => 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,Mac...

...oing some concept testing with FWD for toll free calls, but I am using 393 as a trunk access code. Question: Will Asterisk know that the one Teliax circuit is in use and use a different trunk? How would I make the dialplan to use a different trunk if the Teliax one is busy? Currently I have: [outrt-003-dial9] include => outrt-003-dial9-custom exten => _9.,1,Macro(hoodahek,${ARG1}) exten => _9.,2,Macro(dialout-trunk,1,${EXTEN:1},) ;or could be Dial(Zap/g1/${EXTEN}) ;exten => _9.,3,Macro(outisbusy) ; No available circuits ;Since this is a PRI group, I am not sure how it's i...

I have an analog GSM Gateway that is connected to a normal SIP ATA device. Basically what it does is this : when you call the extension nr. of the SIP ATA port, the GSM Gateway will pick up the phone and presents a (new) dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia a Grandstream HT286. I would like to use the GSM Gateway to route my outbound cellular calls, how

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

...t=On-Demand record_in=On-Demand qualify=no port=5060 nat=never mailbox=23@default host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="SIP Lite" <23> [sip-out] type=peer host=209.XXX.XXX.113 -----------------Extensions_additional-------------------------- [outrt-001-sip-out] include => outrt-001-Prizm-custom exten => _011.,1,Macro(dialout-trunk,4,${EXTEN},) exten => _011.,2,Macro(dialout-trunk,1,${EXTEN},) exten => _011.,3,Macro(outisbusy) ; No available circuits exten => _1NXXNXXXXXX,1,Macro(dialout-trunk,4,${EXTEN},) exten => _1NXXNXXXX...

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...

...ing sound problems when diverting a call using asterisk@home 1.5. I am using the following configuration in extensions_custom.conf, extensions_additional.conf and extensions.conf [custom-Sales] exten => s,1,SetVar(DivertNumber=02XXXXXXXX) exten => s,2,Dial(SIP/116, 15) exten => s,3,Goto(outrt-010-outside3,9${DivertNumber},1) (i have replaced the diverted phone number with XXXXXXXX above) [outrt-010-outside3] it's the context to make outbound calls via SIP trunk The custom-Sales context is used in the following ext-did context for incoming calls, [ext-did] exten => 02YYYYYY...

...;wanpipe2 card 1" HDB3/CCS/CRC4 RED group=1,12 context=from-internal switchtype = euroisdn ;overlapdial = outgoing priindication = inband signalling = pri_net channel => 32-46,48-62 context = default group = 63 Span 4: WPE1/3 "wanpipe4 card 3" HDB3/CCS/CRC4 group=4,14 context=outrt-001-PSTN_E1 switchtype=qsig signalling=pri_cpe ;facilityenable=yes ;callprogress=yes pridialplan=unknown prilocaldialplan=unknown ;priindication = outofband ;overlapdial = incoming ;priexclusive = yes ;pritimer => t200,1000 ;pritimer => t313,4000 ;immediate=yes channel => 94-108,110-124 co...

...on "extensions" Now on the right near the top, click on "Browse" This is where AMP keeps all your extension info. This will be the hardest part because you are going to have to do the identification. Typically, the dial commands are kept near the bottom and start with "outrt" You're going to want to find the name of the outbound route that will be using the cellsocket. Look for the entries for that route that contain "dialout-trunk", those will be the ones you want to edit. Click on the little pencil icon for that line. Go to the "dialout-tr...

...=2405243333 insecure=very context=frombroadvoice authname=2405243333 dtmfmode=inband dtmf=inband In my extensions.conf I have: ;setup SIP extension for BroadVoice [globals] BVNUMBER=2405243333 ; your calling number BVRINGS=201 ; the phone to ring BVVMBOX=201 ; the VM box for this user [outrt-003-BroadVoice] include => outrt-003-BroadVoice-custom exten => _8.,1,Dial(SIP/${EXTEN:1}@sip.broadvoice.com,30) ;exten => _8.,1,Dial(SIP/${EXTEN:1}@2405243333,30) exten => _8.,2,Congestion() exten => _8.,102,Busy() [frombroadvoice] exten => ${BVNUMBER},1,Macro(exten-vm,${BVRING...

...=400 rxgain=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[3...

...=400 rxgain=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[3...

...out, no authentication 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 => _XXXXX...

...with the disconection of your analog lines ;busydetect=yes ;busycount=3 immediate=no ### PRI group=0,11 context=a2billing switchtype=euroisdn priindication=inband overlapdial=yes nsf=none signalling=pri_net channel => 1-15,17-31 context = default group = 63 ### MFCR2 group=2,12 context=outrt-005-IN_E1P2_PBX_JLB signalling = mfcr2 mfcr2_variant=ph mfcr2_max_ani=10 mfcr2_max_dnis=4 mfcr2_get_ani_first=yes mfcr2_category=national_subscriber mfcr2_logdir=span2 ;mfcr2_call_files=yes channel => 32-46 channel => 48-62 context = default I am not sure if I miss anything in the configurat...

...eed to change, if you're using AAH. $strUser = "admin"; #specify the password for the above user $strSecret = "amp111"; #specify the channel (extension) you want to receive the call requests with #e.g. SIP/XXX, IAX2/XXXX, ZAP/XXXX, etc $strChannel = "Local/15555555555@outrt-001-telasip"; #specify the context to make the outgoing call from. By default, AAH uses from-internal #Using from-internal will make you outgoing dialing rules apply $strContext = "from-internal"; #specify the amount of time you want to try calling the specified channel before han...

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

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

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