search for: _5xxx

...,1,Flash() exten => _NXXX,n,SendDTMF(${EXTEN}) exten => _NXXX,n,Hangup() [from-zap] exten => s,1,Dial(SIP/sv0071iv) exten => s,n,Dial(SIP/sv0072iv) exten => s,n,Goto(AA,s,1) [AA] exten => s,1,Wait(.5) exten => s,n,Background(vm-whichbox) exten => s,n,WaitExten exten => _5XXX,1,Playback(transfer) exten => _5XXX,n,Flash() exten => _5XXX,n,SendDTMF(${EXTEN}) exten => _5XXX,n,Hangup() --- Now, for whatever reason if sv0071iv, and sv0072iv fail to qualify), asterisk will play a simple menu choice asking which extension they want to transfer too (Mind the Backgroun...

Good day all I've tried my iax conf and I'm struggling.So I want to know If someone else got this working and if they can pleas send my their configs I have to asterisk server,in different tows,both offices connected wit a direct line so both servers are on the same network running SIP.Each town got different extension register to each sever.Town A=100+ town B=200+ How do I get town A

...ql: +-----+----------------------+-------+----------+----------+------------------------+ | id | context | exten | priority | app | appdata | +-----+----------------------+-------+----------+----------+------------------------+ | 155 | citicomco-internal | _5XXX | 1 | MeetMe | ${EXTEN}|cMrpsq | | 285 | citicomco-ivr-extens | _5XXX | 3 | Goto | citicomco-incoming|s|1 | | 283 | citicomco-ivr-extens | _5XXX | 1 | MeetMe | ${EXTEN}|cMrpsq | | 284 | citicomco-ivr-extens | _5XXX | 2 | Playback | goodbye...

...[2000] type=user username=2000 auth=plaintext permit=a.b.c.d/255.255.255.0 host=dynamic context=fullaccess My extension.conf is as follows for the server that is directly connected to internet.: [fullaccess] exten => _5XXX,1,Dial(IAX2/1000@a.b.c.d/${EXTEN}) exten => _5XXX,2,Hangup exten => _5XXX,102,Hangup -------------- Now the iax.conf file for the one behind NAT is as follows: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on m...

I'm trying to get two * boxes to talk.... no matter what variation I try I get No Authority Found and connection refused from 192.168.1.5 I've googled, I've site searched.... to no avail. Here is the server a configs (192.168.1.5): iax.conf [general] port=5036 bandwidth=low disallow=all allow=gsm jitterbuffer=yes tos=lowdelay register => pbx:test@192.168.2.2 [pbx] type=peer

Hello List. I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one. I would like to run different scenarios: 1. Have one of the boxes at a different location outside the LAN and have them communicate. 2.

Hi guys, i'm trying to use the "featuremap" of features.conf inside the app meetme, but it's no working. like: _5XXX => { Set(DYNAMIC_FEATURES=toca_macaco); MeetMe(${EXTEN},F); //F forces the meetme to pass DTMF Hangup(); }; in features.conf: toca_macaco => 123, peer, Playback,tt-monkeys But, if, inside the room, I press *123* the sound file tt-monkeys it's not executed....

...ecret=xxxxxxx nat=yes extensions.conf [internal_users] exten => 6000,1,Answer() exten => 6000,2,Playback(hello-world) exten => 6000,3,Hangup() exten => 6001,1,Dial(SIP/6001) exten => 6002,1,Dial(SIP/6002) exten => 6099,1,Playback(tt-weasels) exten => 6099,n,HangUp exten => _5XXX,1,Dial(${IAXTrunk}/${EXTEN}) same => n,Hangup() exten => s,1,Answer() exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() IAX.conf [trunk-1] type=friend username=trunk-1 trunk=yes requiretoken=no secret=password host=172.16.200.212 context=internal_users auth=plaintext disallow=al...

...=>Asterisk:XXXXXXX@fedcore2.XXXXXXXX.com/5000 [sip_proxy-out] type=friend ; we only want to call out, not be called secret=XXXXXXXX username=Asterisk ; Authentication user for outbound proxies host=fedcore2.XXXXXXX.com In my extensions.conf I have exten =>_5XXX,2,Dial(sip/${EXTEN:1}@sip_proxy-out,20,r) So that dialing an extension 5XXX rings sip extension XXX. I also the following context to catch incoming SIP calls. [sip-incoming] exten=>s,1,Wait,1 exten =>s,2,Goto(default,384220,1) exten =>5000,1,Goto(default,384220,1) exten =>_9.,1,Goto(d...

Hello all, Is there anyway that I can configure Asterisk to start dialing out from fxo after (xx) seconds from getting the dial tone? I don't want tdm card to send the number immediately because it fails many times. Thanks for any help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, My client makes manual sales calls to prospects. He is often sent to voicemail on the prospect's side. If he finds himself having to leave a message, he would like to be able to press a key and let a pre-recorded message play into the prospect's vmail box. This is so he can maintain consistency in his message. Can anyone offer suggestions of how I could accomplish this

...asterisk-1 where you want the call to go, so it is going to 's'. Try adding the extension to the Dial() command on asterisk-2. Change Dial(IAX2/trunk-1) to Dial(IAX2/trunk-1/${EXTEN}) Note: It appears that you are doing it correctly from asterisk-1 towards asterisk-2 exten => _5XXX,1,Dial(${IAXTrunk}/${EXTEN}) Assuming, of course, that the variable IAXTrunk is properly set. Dale -- "The truth speaks for itself. I'm just the messenger." Lyta Alexander - Babylon 5 -------------- next part -------------- An HTML attachment was scrubbed... URL: <h...

I have googled, and wiki'ed until blue. Is it possible to put T1---*----Toshiba CTX ? I have a TE405P, with one interface programmed for the T1, I am not sure how to program the 2nd port to mimick the T1 to the Toshiba. The Zapata.conf [channels] switchtype=national context=from-pstn signalling=pri_cpe usecallerid=asreceived echocancel=yes echocancelwhenbridged=no echotraining=400

...ssage " 513 Message too big" back from SER. Any ideas what I am doing wrong? Debug below. SER and Asterisk are running on the same Server SER is on port 5060 Asterisk is on port 5061 In my extension.conf I have the line SERADDRESS=192.219.85.57:5060 in Globals and am using exten =>_5XXX,2,Dial(sip/${EXTEN:1}@${SERADDRESS}) to dial out. Here is the sip debug. -- Executing Ringing("H323/ip$192.219.85.57:2680/5746", "") in new stack -- Executing Dial("H323/ip$192.219.85.57:2680/5746", "sip/290@192.219.85.57:5060") in new stack We'...

Yes. Tim Thompson http://www.amatechtel.com (806) 722-2227 -----Original Message----- From: CW_ASN - Gus [mailto:cw_asn@fibertel.com.ar] Sent: Wednesday, October 22, 2003 1:12 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Meetme Do you have ztdummy or zaptel device in your system? ----- Original Message ----- From: "Panny Malialis"

...=>Asterisk:XXXXXXX@fedcore2.XXXXXXXX.com/5000 [sip_proxy-out] type=friend ; we only want to call out, not be called secret=XXXXXXXX username=Asterisk ; Authentication user for outbound proxies host=fedcore2.XXXXXXX.com In my extensions.conf I have exten =>_5XXX,2,Dial(sip/${EXTEN:1}@sip_proxy-out,20,r) So that dialing an extnesion 5XXX rings sip extension XXX. I also the following context to catch incoming SIP calls. [sip-incoming] exten=>s,1,Wait,1 exten =>s,2,Goto(default,384220,1) exten =>5000,1,Goto(default,384220,1) exten =>_9.,1,Goto(d...

...=yes context=default group => 1 stripmsd=> 1 channel => 1-2 ------------------- and here is a portion of extensions.conf which deals with outside call (or tries); exten => 4000,1,Playback(tt-weasels) exten => 4000,2,SetCallerID(340) exten => 4000,3,Dial(Zap/1/333) exten => _5xxx,1,Dial,Zap/g1/BYEXTENSION can anybody help me to fix what I am doing wrong? ---- The linuX Files -- The Source is Out There. mailto:marko@printel.hr http://printel.hr

...for "${EXTEN}) > ;exten => 4XXX,2,Dial(H323/${EXTEN},60,tr) > ;exten => 4XXX,3,Congestion > > [outbound] > exten => _4XXX,1,NoOp("Route to CCM1" ${EXTEN} ) > exten => _4XXX,2,Dial(H323/${EXTEN}) > exten => _4XXX,3,Congestion > > exten => _5XXX,1,NoOp("Route to CCM1" ${EXTEN} ) > exten => _5XXX,2,Dial(H323/${EXTEN}) > exten => _5XXX,3,Congestion > > exten => _9NXXXXXXXXX,1,NoOp("Route to CCM1" ${EXTEN}) > exten => _9NXXXXXXXXX,2,Dial(H323/${EXTEN}) > > exten => _91NXXXXXXXXX,1,NoOp...

Hi, I'am comparative newbie to the world of Asterisk. I'd like to introduce an Asterisk based PBX into my company but need to convince my executive of it's worthiness. I need some reference sites to quote in my discussion, preferably well known companies of course. I have surfed the net but not come up with anything of note, if anyone can help it would be greatly appreciated.

...llow=ulaw ; Allow codecs in order of preference register =>Asterisk:asterisk@fedcore2.eicon.com [sip_proxy-out] type=peer secret=asterisk username=Asterisk fromuser=Asterisk host=fedcore2.eicon.com dtmfmode=inband > In my extensions.conf I have > > exten =>_5XXX,2,Dial(sip/${EXTEN:1}@sip_proxy-out,20,r) > > So that dialing an extension 5XXX rings sip extension XXX. > > I also the following context to catch incoming SIP calls. > [sip-incoming] > exten=>s,1,Wait,1 > exten =>s,2,Goto(default,384220,1) > exten =>5000,1,Goto(d...