search for: lamineka

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

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 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

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.

Hi everybody, I would like to know how can I detect events in queues. For example when an operator answers a call which was in a queue I would like to some informations related to the caller (quote from the database) to the operator. I saw these events in queuelog.conf but i don't know how to retrieve them in my AGI. Thanks for any help Lamine

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 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

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, Does someone out there ever heard about PRAD or V5.2. Is there any link with Digium's TE110P? Thanks for any enlightments Lamine

Hello, I would like to configure a queue with only one agent. But while waiting the callers in the queue should listen to the agent's conversation instead of a music on hold. Is that possible with Asterisk? Does someone out there ever work on a similar configuration? Thank you for any suggestion Lamine

Hi All, Is the astmanproxy yahoo group (http://tech.groups.yahoo.com/group/asterisk-astmanproxy/) still working? It seems to me the most recent posts are 2006's. I have sent a message but didn't receive any feedback and the post was not listed. Is the project still being maintained? Is there any other discussion forum about? Thanks for any update Best regards -------------- next part

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

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001