search for: 1234,3

I'm running 1.2.9.1, and I can't get caller id dialplan matching to work. When calling from 9220370 to 1234, the following does not match. exten => 9220370/1234,1,NoOp(${CALLERIDNUM}) exten => 9220370/1234,2,Answer exten => 9220370/1234,3,Playback(tt-weasels) However, when calling from 9220370 to 1234, this DOES match. exten => 1234,1,NoOp(${CALLERIDNUM}) exten => 1234,2,Answer exten =&...

...directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty application=http://pubint.ic.llnwd.net/stream/pubint_wnpr and this is how I am testing it: exten => 1234,1,Answer exten => 1234,2,SetMusiconHold(stream2) exten => 1234,3,WaitmusiconHold(60) exten => 1234,4,Hangup and this is the console output I get when I dial 1234: Asterisk Ready. *CLI> -- Executing Answer("SIP/3250072-ed28", "") in new stack -- Executing Se...

...;nicer' if it would evaluate it this way. (Ref: http://bugs.digium.com/bug_view_page.php?bug_id=0000401 ) If you put a 0 after the } it does work as I would want it to without an error. The other suggestions did not work. I propose for this bug to be re-opened. extensions.conf: exten => 1234,3,GotoIf($[${a}]?4:5) If a is undefined: WARNING[37910]: File ast_expr.y, Line 346 (ast_yyerror): ast_yyerror(): syntax error: parse error -- Executing GotoIf("SIP/1240-5eb6", "0?4:5") in new stack -- Goto (default,1234,5) If I change the extention to exten => 1234...

...ist - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Streaming Music On Hold > > > Try this: > > musiconhold.conf: > > [stream2] > mode=mp3 > directory=http://pubint.ic.llnwd.net/stream/pubint_wnpr > > > extensions.conf: > > exten => 1234,1,Answer > exten => 1234,2,MusicOnHold(stream2) > exten => 1234,3,Hangup > > > On Wed, 2006-02-22 at 09:28 -0700, Douglas Garstang wrote: > > Ok, I'm tearing my hair out trying to get Asterisk moh streaming to > work. After several hours jerking around with iceca...

Hi, I have a question of using 2 SIP providers. Let's say I have provider A and provider B, and I would like my calls to go to A, and then B if A wasn`t available Something like this would work: exten => 1234,1,Dial(SIP/providerA) exten => 1234,2,Dial(providerB) exten => 1234,3,Hangup But what if I want to put in a delay? If I put 30 seconds on each of them, I'll wait a total of 60. I want to wait only 30 seconds before the hang up. Also, if ProviderA has a main server and a backup server...

...tingcallerid=no threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no switchtype=5ess context=main signalling=pri_cpe group=1 channel => 1-23 channel => 25-47 exten => 1234,1,Wait,1 exten => 1234,2,Answer exten => 1234,3,SetCallerPres(allowed_passed_screen) exten => 1234,4,SetCIDNum(8881234567) exten => 1234,5,Dial(Zap/g1/18887654321,,,) exten => 1234,6,Hangup Thanks, Shaun Tierney

I'm trying to set a dialing rule in my dialplan. As a part of it, from my point of view, this works wrong priorityjumping=no [test_context] exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag exten => 1234,2,Playback(digits/2) exten => 1234,3,Playback(digits/3) exten => 1234,102,Playback(digits/4) In this case, if I dial the extension, and it's answered, the caller can hear "Two, Three) and the ca...

There are often times that I want to read a DB value from the dialplan, and if this family/key pair does not exist, set it to some default value. for example: 1234,1 => Set(EMAILADDR=${DB(x/y)} 1234,2 => GotoIf($["${EMAILADDR}" = ""]?3:4) 1234,3 => Set(EMAILADDR=Someone@test.com) 1234,4 => NoOp(${EMAILADDR}) 1234,5 => Hangup() I have modified the db function to take an extra parameter to set if the key does not exist. So,...

Hi All, I'm having trouble setting up a queue: I'm using AgentCallBackLogin to login in the queue, but: 1 - When an agent answer the call and another call arrive his phone rings again. 2 - When no there are no one answer the queue the system goes to voicemail of agent 1234 I'm using asterisk-1.2.0-beta1. My configuration is below, Any ideas? Many thanks, Joao Antunes ;extensions.conf [demo] exten => 1005,1, Answer exten => 1005,2, AgentCallBackLogin(${CALLERIDNUM}|${CALLERIDNUM}@agentes) exten => 1005,3, AddQueueMember(test1|local/${CALLERIDNUM}@age...

Does anyone have an example of hoe to use these two commands? I have read he documentation, and I am still unclear on where this command goes, as part of extensions.conf or where? If someone could post a working example it would be most helpful. Regards to all, Joe

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the

...setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based - no real phone lines). I tried this (from voip-info.org wiki)... exten => 1234,1,dial(sip/1234,20) exten => 1234,2,playback(pls-wait-connect-call) exten => 1234,3,Setvar(NewCaller=${CALLERIDNUM}) exten => 1234,4,SetCIDNum(0${CALLERIDNUM}) exten => 1234,5,dial(${TRUNK}c/9871234321,20,r) exten => 1234,6,SetCIDNum(${NewCaller}) exten => 1234,7,voicemail2(u1234@...

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

I've had my * setup installed with an X100P card for a couple of weeks and it's great fun! I'm even giving a demo to the local Linux group in a couple of days. But I have a snag. I have the X100P on a shared line, and configured to wait for 20 seconds before answering and doing the auto-attendant/voicemail dance. My problem is I can't find an application command to cancel the

...these routines are also coming in via SIP. Voice works fine on the channels, once answered. Does anyone know if there is something wrong with the "r" modifier on SIP Dial application calls, or have you had experience doing this a better way? None of these methods work: exten => 1234,1,Dial(${PHONE1},25,r) exten => 1234,1,Answer exten => 1234,2,Dial(${PHONE1},25,r) exten => 1234,1,Answer exten => 1234,2,Ringing exten => 1234,3,Dial(${PHONE1},25,r) exten => 1234,1,Ringing exten => 1234,2,Dial(${PHONE1},25,r) exten => 1234,1,Ringing exten => 123...

Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360?

Hi, alhtough i googled for details concerning ASTCC i did not found an aswer to the following: Should i put in my extensions.conf the configuration of the astcc? I ask this because as i see it, in the end of the extensions.conf there is an include statement : #include /var/lib/astcc/astcc-exten.conf Should the config been done in the astcc-exten.conf file or the initial extensions.conf

...0.0 > > txgain=0.0 > > group=1 > > callgroup=1 > > pickupgroup=1 > > immediate=no > > switchtype=5ess > > context=main > > signalling=pri_cpe > > group=1 > > channel => 1-23 > > channel => 25-47 > > > > exten => 1234,1,Wait,1 > > exten => 1234,2,Answer > > exten => 1234,3,SetCallerPres(allowed_passed_screen) > > exten => 1234,4,SetCIDNum(8881234567) exten => > > 1234,5,Dial(Zap/g1/18887654321,,,) > > exten => 1234,6,Hangup > > > > Try something like th...

Hi, one short question: Is it possible for the zaptel driver to deal with multiple phone numbers on one single E1 PRI line? I could make my carrier route +49 xxx aaaaa-zzz and +49 xxx bbbbb-zzz and others down one single PRI trunk to our asterisk box terminating in a Digium TE410P. Does the driver handle this and can I put calls coming in all on the same physical interface put into

I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael