similar to: Listening to agent's conversation while waiting in the queue

Hello everybody, I would like to know if there is a support of IAX in vicidial. I want to make predictive dialing use vicidial using IAX soft phones. Thanks in advance Lamine -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040924/d1cc487b/attachment.htm

Hi everybody, I would like to decrease the load of my asterisk server. Could someone recommend me a solution? I have thought about a hardware component that would do some tasks as compression/decompression or codec translations but wonder if such a solution exist. Thanks for any suggestion Lamine

Hi everybody, I have been searching around for days on how to record calls between SIP phones.Could someone tell me whether it is possible? The Record command doesn't seem to work during a call. Thanks Lamine

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close

Hi everybody I am having problem detecting fax with my X100P. I have RedHat 8 as OS and an X100P and a TDM400P. The X100P being plugged into PSTN. I have successfully installed tiff-v3.5.7 and spandsp-0.0.1 and also patched Asterisk wthout problem. Here is my zapata.conf file context=cda signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=1.5 txgain=1.5

Hello everybody, I have an Asterisk box with a TDM04B and would like to connect it to a GSM Gateway. Can someone tell me whether i can get the callerid for incoming calls in this case? Thanx Lamine

Hi everybody ! I had patched asterisk to install chanspy weeks ago and everything was ok. With the current version of cvs i am having failures when i try to apply the same patch and the url where i originally downloaded it seems no longer active. Is the patch any longer maintained or has it been replaced with another function. Thanks in advance Lamine

Hello everybody, I would like to know whether it is possible to run Dial and MeetMe commands simultaneoously on the same channel. I am using a C AGI as below but it seems to me that only the first command that is called in the agi is executed. ........... // Pr?paration de la commande pour l'appel du client fprintf(stderr,"%s%s",numtocall," is the number to

I'd like for our custom soft phone to be able to know what queue, and/or what DID is calling an Agent's phone before the agent picks up. The agent is using the AGENTCALLBACKLOGIN. One agent can be in multiple queues so it would be nice if they could get a pop up window telling them who's on the line before they pick up and hear the announcement telling them that. I'd like

Hello everybody! I have been using MeetMe application with SIP and Zap devices and everything worked fine. Now I want to use an IAX client but when trying to connect to a conference I get a message telling that the conf number is not valid. Through documentation it seems to me that conference should be enabled for IAX clients but I don't know how. Please can someone tell me? Thanks in

Hi, Is it possible to monitor conversation of logged in Agents? Currently I am using ZapScan to monitor incoming calls, but I would like to monitor individual agents. raj

Hi!, I'm trying to set up a Queue (which works fine now :-) Sip clients can login in to the Queue with dialing 91 on there phone. And as soon as there are customers the Queue calls the agents back. I would like that the queue calls the agents also if it's phone is call-forwarded. With agents (sip clients) are added with the following extensions: exten => 91,1,AddQueueMember(myqueue)

Hi everyone. I'm a bit of a Linux newbie, but I've been doing tech stuff for ages. I'm also brand new to *. I've been reading the Voip.org wiki, and perusing the list archives for a while since I've been asked to investigate using IP telephone / soft phones for a call-center type scenario. People (marketing folks) have pointed me at Cisco, but I really don't wanna.

i'm planning to migrate a call center to asterisk, i don't understand if i can launch a resident application on the agent's client in relation with the queue the agent's is answering. For example: I have - queue A - queue B - queue C Agent 100 (logged in A.B,C) Agent 101 (logged in C) When Agent 100 receives a call from the queue A i'd like to launch his browser and point

Hi. Today I was asked about a function of asterisk. That's what it should to: a call arrives -> put it in a queue -> remain here ;) Then, when someone wants to answer, just dial an extension and the older call that's in the queue is picked up. A sort of app_queueonly + app_pickolderqueuecall . As far as I know asterisk doesn't support that, so was wondering if someone

http://bugzilla.mindrot.org/show_bug.cgi?id=480 Summary: extend the -l or -L option to mention the ssh-agent's pid Product: Portable OpenSSH Version: older versions Platform: All URL: http://www.openbsd.org/cgi-bin/man.cgi?query=ssh-add OS/Version: All Status: NEW Severity: enhancement

Hi All, I have a question about the Queues. I'm using Asterisk 11.13.0 , and I want to configure the following setup : When there is an incoming call to the queue all agents should ring even those that are already in call, they should receive a second call. Is this doable in any Asterisk version ? Thanks in advance. -------------- next part -------------- An HTML attachment was

<asterisk-users@lists.digium.com>Asterisk 1.4 I have strategy= leastrecent and autofill = yes I have 2 agents, one is answering a call and the other is free and have some calls waiting in the queue. Only when the first agent hangup the second agent receive the first call in the queue. It happends some times. This behavior still happend in 1.4.1 version. Thanks a lot. -------------- next

While we wait for queue to finish draining, use completions instead of uslee_range(). This is better way of waiting for event. Signed-off-by: Vivek Goyal <vgoyal at redhat.com> --- fs/fuse/virtio_fs.c | 39 +++++++++++++++++++++++++++++---------- 1 file changed, 29 insertions(+), 10 deletions(-) diff --git a/fs/fuse/virtio_fs.c b/fs/fuse/virtio_fs.c index 43224db8d9ed..b5ba83ef1914 100644

Hey all! I'm trying to figure out how to have a queue accept an inbound caller's key press to action on. At first I'm just trying to implement a "Press 1 to leave a voice mail" announced and at any time in the queue, the user can press 1 and go to the queue's voicemail. Later I'd like to have it accept "Press 1 if this is an x issue, press 2 if this a y