similar to: Converting GSM calls to SIP

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:

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, 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.

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

Anybody notice log delays in this list, and very small amount of traffic? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/576a9b0e/attachment.htm>

Does anybody know of a Similar list for SwitchVoX? And would like to post to proper list if one is available. I had posted on digium forum, but have not received any responses yet. http://forums.digium.com/viewtopic.php?f=38 <http://forums.digium.com/viewtopic.php?f=38&t=77031&sid=4adb81c464701e0039d e21a300aa273f> &t=77031&sid=4adb81c464701e0039de21a300aa273f

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

Hi, I know you can access various sip variables via 'Set(sstatus=${SIPPEER(201,status)})' (for example) to get the status of the sip user - but what about variables? I have a user that has setvar=123456 in their users.conf (sip.conf if you prefer). I can read it with a 'sip show peer 201' - but that gives everything and parsing that isn't really an option. Anyone know how

Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error "FAX handle 0: failed to queue document 'filename.tif'", so we set it to 1680x2285, but it's still rejected. Is there a way

I'm working on getting the latest Linux distros working well on one of our prototype machines, however, some of them are failing to boot into the installer from the CD/DVD images. I've narrowed it down to isolinux hanging just before displaying the graphical menu. After a little bisecting between the last version that worked (3.73) and the first version which was broken (3.74), I found

Hi, my Asterisk on debian lenny died after 80 days. server kernel: [7572666.186852] asterisk[3673]: segfault at 10 ip 7f3b8e90b4aa sp 40bf5f00 error 4 in l ibpthread-2.7.so[7f3b8e903000+16000] Anything what can be done to find out the reason? best regards Thomas

Hello, I have a Google Voice phone number and want to connect it to my asterisk box to have calls handled to my SIP account. When I call the number I receive the correct INCOMING request on Jabber portion of asterisk, but the call is not connected to the gtalk part. JABBER: asterisk INCOMING: <iq from="+ 17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4" to="

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 am trying to determine what would need to be done/modified to enable the following: I have a SIP extension come into my asterisk box, and I then need it to call "6-10" remote Sip Stations that are set to Auto-Answer... (note, my remote sip stations are actually cisco h323 devices, I can call them fine from any softphone, or other device, and have full-duplex audio, however, i need to

I work for Terra Soft Solutions (makers of Yellow Dog Linux) and I'm trying to compile the latest cvs snapshot for inclusion in our next release, but I'm running into some problems... I'm using modified versions of the SRPMs included in RedHat's Rawhide distro (I only updated to the latest cvs, but the old version also experienced this problem), which compile fine on an x86 box I