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Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Hi Guys, i am using two PBX's i can call from pbx1 a cellphone tied to pbx2 that way: 1) use a phone in PBX1 2) call extension in PBX2 3) extensions in PBX2 ring to a cellphone (as this specific ext is tied to a cellphone) my questions now is : am i gonna be able to dial from an IPphone registered within PBX1 to a cellphone by using the Trunk (Zap or dahdi) of PBX2? anybody know

I have several remote phones that experience a slight "call" delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have asked a GSM operator in my country if he can route a number or a short code to my asterisk server via SIP (since they dont give DIDs in my country) the operator said they do not support SIP, they have no way of converting GSM calls to SIP to then send them to me. I would like to know what is needed from the operator side to do this, what kind of material is needed, or what can be done from

Hi, Having problems with getting either RxFax or FaxReceive to compile. Running Asterisk 1.4 on CentOS 5. Does anyone have the free/open source executables that you could send me? Thanks for your help! P. S.: TxFax and FaxSend would also be appreciated.

Anybody know a reliable fax solution for 1.4.30 branch? I am using PikaFax on another server and works very well (about 3000 faxes a week), but it appears they no longer offer their product to open source asterisk, only for there "WARP" appliance. NOT really looking to migrate from 1.4.x to 1.6.x -------------- next part -------------- An HTML attachment was

Hi All, i want to make an extension from pbx1 able to tlak to another extension from pbx2 or use pbx2's trunk to dial outside calls. so i edited in both servers accordinally the iax.conf: register => pbx1:pass at 172.16.200.175 <pbx1%3Apass at 172.16.200.175> [pbx2] type=friend host=dynamic trunk=yes sercret=pass context=[default] ; i used the biggest context to avoid confusion as

Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have several sip stations that on a that are on a nat'd network behind a nice friend firewall.. no audio path issues, 2 way audio works, etc,etc,etc. However, I can't get any of my phones to Transfer or Blind Transfer.. I search and search, and well, just about gone nuts on this one. Here is sip debug from pressing "transfer->blind->dial dest->Dial Key" (note both

Hi Everyone, I am accustomed to PUTTY and it's very nice as in it allows many many SSH profiles to be saved and allows tunneling etc....but it's not very good when it comes to scrolling up and down, colors, text size, and specially it doesn't give a title to the opened instance. Maybe giving the IP address as the title of the window would help a lot if you have many different servers

What is the easiest method or can someone point me in the direction I need to look to do remote agent login.. Ie, Caller calls in with a cell or home phone, authenticates himself, select a queue to be added too, hangs up, and then any calls coming into said queue would ring their home or cell phone, and then they can then callback, and logout from the queue? I see AgentCallBack was

For those who don't know, (as I just figured out by reading the sourcecode), that all settings for a particular "channels" must be placed before the channel => entry. Ie, Immediate=no Channel=>1-24 Immediate=yes Channel=>25-48 Immediate=no Channel=>49-72 1-24 will have immediate set to no, 25-48 yes, 49-72 no Maybe someday the config will be

I know in v1.6 its part of logger.c but I noticed this: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=11625 However, it doesn't seem to ever been applied to any version of 1.4.x branch.. Nor can I figure out what it was applied to? This is over 3 years old, you would of figured it would have been applied to 1.4 at some point in time.. Any ideas?

I am trying to configure a channel bank with 24 ports of FXS., but appear to be hitting a roadblock? This worked on v1.4.xx but now just get "SimpleSwitch" and immediate=no/yes don't seem to make a difference?, no matter if under top section, under channel, etc. Chan_dahdi.conf: [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes