similar to: Asterisk : using setvar with IP Realtime and variable inheritance

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do

Hello, I've got my dialplan configured to do a double ring when a customer service call comes in, and a normal ring when an extension is dialed directly. When a customer service call is transferred, I want to ring to revert back to normal. In the local extension macro, I have the following ; make sure ring is set to default exten => s,n,NoOp(${ALERT_INFO}) exten =>

Hi Guys, Is it possible to harvest the output of system into a SetVar(variable)? exten => s,n,SetVar(var=system(*asterisk -rx "sip show channels" | grep -c "(ulaw)")* * * *??? any problem with the syntax? * * * * * *Thanks,* * * -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a limitation to the number of variables you can set from a PHP agi script? I have a simple example and I can't get it to let me set more than 1. I am pretty sure I am just missing something, but I've searched all over an can't find the answer. Here is the extensions.conf part: exten => _XXXXXXXXXX,1,AGI,diallocal.agi exten => _XXXXXXXXXX,n,NoOp(${ISLOCALCONTEXT})

Would someone be able to give an example of a working use of setvar from chan_dahdi.conf? I am trying to create a custom variable like I use in sip.conf but I have been completely unsuccessful getting any variable set using setvar to appear for a DAHDI channel. I am running 1.8.11-cert8 and am using the newer format (but I have tried using the older [channels] format). Here is an example:

I am trying to have the ring types different for internal and external incoming calls. I have followed the guide on the wiki, the SetVar executes, in extensions.conf I have it as s,1, Yet it doesn't work? When the phone rings, the ring type is the one I chose on the phone, it rings same tone for both when I test. Using Asterisk Stable. Anyone got this working and can

Hi all!! Does anyone have a short example or even better - a working AGI script that uses "GET VARIABLE' from a /var/spool/asterisk/outgoing call that uses "SetVar"? Here's what I've tried with no luck so far: sample.call ================= Channel: SIP/1000 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Application: Agi Data: playTasks.agi Callerid: Nightly Processor

I had lines 3 and 4 and added line 1 and 2 to extensions.conf exten => 106,1,SetVar(CALLFILENAME=${TIMESTAMP}_${CALLERID(num)}) exten => 106,2,Monitor(wav,${CALLFILENAME},m) exten => 106,3,hint,SIP/106 exten => 106,4,Macro(stdexten,106,${HINT}) I received this warning: WARNING[31463]: pbx.c:1832 pbx_extension_helper: No application 'SetVar' for extension

How can I use the column "setvar" in my dialplan? I am not sure if it is for that what I need: Many phones have the same jump in place, but need a few variables different, like tariff, silent, need_password, I have for tariff = 4 variations, for silent=2, for need_password=2 ... If I solve it like now, I need 4x4x2 = 32 context variations. If I could use a field in the Real-time

Hello, I have a couple of 7960 and a quad T1 card on my asterisk box. I want to let the phones to use g729 when they "talk" to each other, but to use g711 when I'm going to route the call out of my network using the T1 card. Everything works just fine between the phones, but in order to be able to make calls through T1 I have to disallow the g729. For this purpose I have the

Hi, What is the syntax to set more than one variable in the SIP.conf file for a particular sip peer? (using the "setvar" line) Regards, Mick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080512/85ce8f28/attachment.htm

Hi, does anybody know about the setvar option in asterisk's sip.conf. I am trying to define it for a peer that's used when making calls using the originate ami call, but it seems to not have any effect. Marcus -- Marcus Hunger - hunger at sipgate.de Telefon: +49 (0)211-63 55 55-61 Telefax: +49 (0)211-63 55 55-22 indigo networks GmbH - Gladbacher Str. 74 - 40219 D?sseldorf HRB

Greetings, I have been using the bcmxcp_usb driver heavily over the last two years as part of a very successful product, first from NUT 2.4.1 and now from NUT 2.6.0. Now I have a need to do setvar of the outlet.n.delay.start variables, so recently I modified bcmxcp.c to do that. I've tested it and it works well for my purposes. My plans are to begin shipping that modified driver as part

Hi all, I run an Asterisk 1.4.24.1 and face problem with DTMF. When calling from my mobile phone -Nokia E65- in GSM, Asterisk present me a second tone so I can use the GW. For this I use: exten => s,1,NoOp(One of our workers (${CALLERID(number)}) is calling office) ;callerID is the one of the calling mobile phone exten => s,n,Background(silence/1) ; Nokia E65 send digits in

Trying to do something like this in the sip.conf under my incoming provider profiles: setvar=CDR(accountcode)=${EXTEN} It seems to show up in the CDR but it's showing up exactly like this "${EXTEN}". Is there a way to stuff the DNIS (number dialed) into the accountcode for CDR? I have already accomplished setting on a number by number basis, I just want to do it globally for all

Hi, I'm currently trying to originate a call with 2 variables set. I tried doing it via manager API and call File and both failed, because the vars were not separated. I'm using Asterisk 1.2_beta1 on this machine Can anyone here verify wether this is a bug or just a stupid error on my part? This is the callfile I tried to use, after the manager way failed: Channel:

All, I have been fighting with my dialplan for hours now, and google searches talk lots but offer nothing in terms of explication for this. I have my SIP peer set up and working with Broadvoice: [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=5555551234 secret=password defaultuser=5555551234 insecure=port,invite context=broadvoice

Hi All, I'm running 1.2.9.1, is setvar= implemented in iax.conf in a later version of asterisk? If so, which one? Thanks. JR -- JR Richardson Engineering for the Masses

Hi I'm using the following sample.call Channel: Zap/2 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: fromfax Extension: 3 Priority: 1 SetVar: num=777777 in extensions.conf I have: [fromfax] exten => 3,1,Wait,1 exten => 3,2,SayDigits(${num}) exten => 3,3,Hangup But it only says 10 it seems variable is not set do I use correctly SetVar ? How can I read variables assigned

Hi all. I'm trying to use the setvar field in my RT sip configuration to store some information about a phone. This works perfectly for my Polycom phones. For the most part, it works for the Grandstream as well. However, when the GS tries to transfer a call, the variable isn't available. For example, I've got a setvar that looks like: setvar line_id=xxxxx If I have an