similar to: Problems with "r" modifier in Dial - does not work in SIP channels?

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

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 =>

Hello Everyone. I usually find my own solutions for problems but this time, after several months, I've given up. My asterisk is set up so that incoming calls from my voip provider ring on both my sip extension and my cellphone at the same time. When the system receives an incoming call, ringtones indicating that the call is being connected play normally for the first 5 seconds to the

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

Thanks to this thread, we got it working too... but have a question... Once this is setup... does it stream forever, or does the stream only start when someone goes on hold/into a queue/etc? If it streams forever, at 24k... it looks like over 7GB/month in bandwidth... so we're not going to want to do that if a) it streams constantly and b) my math is correct. Thanks, Doug >

I am unable to get my 9133i to register with my asterisk server. I am including config files below, this a simple test network so there's nothing secret in the config files. I have upgraded the phone to the latest software version (1.4.3) I'm not sure what the problem is. I can call the phone from a softphone, but the 9133i says "no service" on the screen and I can't dial

Hi, I have an X100P which receives an analog line from another PBX. These are the relevant entries in extensions, PHONE1=Zap/1 [macro-extensions] exten => s,1,Dial(${ARG1},20) exten => s,2,Voicemail2(u${ARG2}) exten => s,102,Voicemail2(b${ARG2}) exten => s,103,Hangup [home] include=>tozap exten => 2201,1,Macro(extensions,${PHONE1},${PHONE1VM}) exten =>

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

I was playing around with conferencing tonight. I was able to place a bunch of SIP phones and a couple of my Zap FXS phones into a conference. So I thought, "Let's see what it's like when people come in from outside." So I called a friend and had him call in on one of my Zap channels, WHICH IS CONNECTED TO MY POTS LINE THAT DOESN'T DO DISCONNECT SUPERVISION. When he

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system 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

hi folks. is there anyway to make the dial command return and execute the next line in the dial plan after the channel hangs up? suppose i want to do something like this: exten => 1234,1,dial(SIP/1234) exten => 1234,2,<do something> but when the dial command hangs up normally, line 2 won't get executed. -- Edwin Lam <edwin@officegeneral.com> Systems Engineer, Office

As far as I can tell, after discussing this matter with other asterisk users in my area, my telco _does_ provide disconnect supervision.. It seems that the problem is actually related to the Sangoma A200 card I'm using, as two other people both using this same card have expressed the same problem.. Are there any other users on this list using the Sangoma A200 FXO port card, and experiencing

Hi, I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has excellent voice quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now. I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as

Hi all, Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use [from-sip] exten => 7723,1,Dial(SIP/897224@fwd) and this works whereas when I'm inside agi app, $AGI->exec('Dial',"SIP/897224@fwd") and this DOESN'T work. There some errors about invalid argument. If I were to do

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

Hi Everyone, Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Even though FreePBX Inbound has an option for Alert_INFO but that doesn't work when the call comes into an IVR or Queue. The calls has to go directly to extension for external ringtone to be different. So, I am looking for internal calls

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I'm a bit confused about how to handle this. I have Asterisk sitting in the middle between a Qwest Long Distance T1 (Voice T1, D4, SF, AMI) and an external voice mail PC using a Dialogic D/240SC-T1 card. The Qwest T1 originally was connected to the Dialogic card directly. The signaling was set to E&M Wink Start because Dialogic used this as its default settings, so it just worked

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re: