similar to: Call-pickup function in Queue application

Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to <sip:bla@voiphost> from "Bla

Hi all, I'm trying to make a context that will monitor a call and when it's completed it would e-mail the wav to a specified mail adres. So I made a standard context that records a call, like this: exten => _*31*00[1-9].,1,Setvar(CALLFILENAME=CALL-${EXTEN:4}-$ {TIMESTAMP}) exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m}) exten =>

Hi All, First off all, this is my first mail to this mailing-list, so if I am doing something wrong please tell me. And apologies for my english in advance, it's not my native language. Anyway, I have few machines running Asterisk 1.2.7.1. All machines but one are Gentoo (other one is Debian). The problem is that Asterisk keeps eating my memory. Just random (mostly at night) all my free

Hi All, I have a SIP provider that tells me that my RTP stream uses a "20bytes payload in the g729 coded data". And they would like that we change this to 30bytes (3 frames). But maybe I'm wrong but isn't a certain payload just a standard for a codec ? And if I'm wrong, how can I change the payload for my g729 calls in Asterisk. Greetings, Attilla

Question: Is it possible to pickup someone else's call who didn't "park" a call? My boss would like to see a way to pick up some one else's incoming call if they aren't at their desk and it's not forwarded. So if my phone were ringing and he knew i ran down the hall, he could press some key combo and give my extension, and it would transfer that incoming call to him.

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks

I have just started to play with callgroups and pickupgroups.. I updates my * from CVS this morning (about 15 mins ago).. I have placed callgroup=1 and pickupgroup=1 into each of my 3 phone configurations in sip.conf.. I place a call from phoneA to phoneB, then I go to phoneC and dial *8# , the call does not get picked up by phoneC and continues to ring on phoneB.. Have I not configured

callgroups & pickupgroups greater than 31 are not working for sip calls with 1.2.14 tarball. Anyone know which branches support 64? John

Hi I have an array in a helper method and I want to oupt the contents to the screen, so i can see waht array key''s are avalible How can this be done? -- Posted via http://www.ruby-forum.com/.

Hi all, at time i am trying to get a better idea of callgroups and pickupgroups (especially within the SIP Channel) A Pickupgroup is relative clear - everyone in the same pickupgroup may pickup a call And a callgroup does what ? - The same ? I thought that a callgroup would act like the ZAP groups - so that you then can dial SIP/g1 - and every SIP Client which is in the callgroup 1 does then

Hi, I have been trying to workout how to use the call pickup. So Far, I have the following in zapata.conf [channels] signalling => fxo_ks context => local pickupgroup=1 callgroup=1 channel => 1-3 When I dial *8# all I hear is busy tone. What have I missed? thanks Jay.

There is plenty of information on the wiki for setting asterisk up for transferring calls both from the Dail() command, and features.conf. What really seems to be missing, is simply how do you actually perform the transfer? Blind transfers are pretty simple as you only have two obvious steps. How though do you do attended transfers? 1.) You have a call 2.) You dial *2 or whatever you have

Should setting a namedcallgroup & namedpickupgroup supersede numeric callgroups and pickupgroup ? I've got 5 peers on my 13.7.0 box, Three of them have a namedcallgroup & namedpickupgroup of 'kiniston' and Two of them have a namedcallgroup & namedpickupgroup of 'sanday'. I'm not specifying a numeric callgroup or pickupgroup so all the peers are defaulting to

Does asterisk have a command to pickup another ringing extention? I've tried searching but couldnt didnt anything. Kyle -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040420/952ce2d4/attachment.htm

It there avalible quality and delay test for sip connections for asterisk. Something like to clients making a call with different codecs and measuring delay , jitter ? I know there is a Astertest but in that you need 2 asterisk mashines (which is usually hard to have). I was looking for perl/bash scripts running sip clients in a finite loop + etheral to measure packet properties , gathering logs.

Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. from my dial plan: [incoming] exten => s,1,Dial(SIP/somebody1|60|tTrR) [internal] include => outbound-local include => parkedcalls

>A "rough and tumble" player that can be distributed with content to >allow for those people lacking a more functional player isn't a bad idea >at all. > Thanks. > There is a lot of complexity associated with it, though. Even if >you elect to just have a command-line player, there's still the audio >interface to deal with. > >Not impossible, just

Hello, when picking up an incoming call from one ip phone on another ip phone, the call terminates after about 5 to 10 seconds. When reading out the hangup cause variable in the h-extention of the dialplan, the hangup cause seems to be 111. In the dialplan output, you can see that SIP-peer sipacc3 picks up the incoming channel SipAgenT01-00001454, and the call is answered. After 7 seconds,

>I don't think a player is what is required as much as hooks into existing >applications/players. > While those are good too the main purpose of a stand-alone player would be something that people could distribute with their files(which is why it should be small). When I say cross platform I mean that it could easily be ported and/or have as similar an interface as all the other