similar to: Setvar SIP_CODEC

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

Hi, I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec being used. I have exclusively Polycom phones for this test, and basically I want all communications to use g729 (preferred codec), except for pagine 20 phones (which busts my g729 license count). In that case I want to use gsm. I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the appropriate

I am wondering if the SIP protocol and its implementation in * allows for changing codecs mid-connection. I've seen some questions regarding this on the list, but I've not found any clear answers. I've also seen the SIP_CODEC variable, but it's not clear that it will change the codec on an existing call. Also, there are mentions of needing a reinvite to make the change, but most

Hi, Is it possible to set a variable for a context for all extensions? I haven't been able to find it. I want something like this in extensions.conf: [from-iaxfwd] exten => .,1,RING=r3 exten => 123456,1,Goto(from-pstn,s,1) [from-internal] exten => .,1,RING=r2 include => ext-local [ext-local] exten => 1,1,Dial(Zap/1,${LONGTIMEOUT}) exten =>

hi: when using chan_sip, I can use set SIP_CODEC in dialplan to change the codec of endpoint. this method didn't work with pjsip in asterisk 12/13. I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems can set codec, but the document didn't offer any example. i try to use something like PJSIP_MEDIA_OFFER(alaw) but didn't work.

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 =>

hello: i want to test the g729 with asterisk. my scenario is sipp(ulaw)->asterisk1 with g729->asterisk2 with g729. I want to test g729 module with asterisk-1.4.21, when i make calls from asterisk 1 to asterisk 2, the asterisk 1 always send ulaw to asterisk 2. my sip in asterisk 1 is with codec g729 and enforce that use g729, the sip in asterisk 2 also work with G729 only, but asterisk 2

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:

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

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})

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 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:

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

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

Hi, I have what I think is a relatively advanced question. Any help is appreciated, even if it's not a complete answer. I am using Asterisk in mostly realtime fashion, specifically SIP registrations are in a MySQL table. This works fine (mostly). I also set a few variables in the setvar column, like this: callerid_internal=test <710>;did=5555551234 Again, this works

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

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