similar to: dropped calls from queue

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

-----Original Message----- I'm crazy here trying to make X-Pro use ONLY g729, and you're struggling to make it not to use it :)... Can you please indicate what's your config for X-Pro and sip.conf? sip.conf: [12345] type=user username=12345 secret=12345 nat=no host=dynamic reinvite=no canreinvite=no disallow=all allow=g729 allow=g729a allow=g723.1 allow=g726 allow=ulaw allow=alaw

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

>> We're having a problem with transfering calls. Our channels are not the >> same as the extensions. We use words instead of numbers. So our config >> looks like this: >> >> SIP/HRUTTER, 1, "81101 Hildegard" >> SIP/JFOLEY-GS, 2, "81103 Jerry" >> >> Consequently when I drag and drop to transfer a call

http://sip.house.com.ar/operator Its a server/client combo that displays the status of your Asterisk PBX in a web browser in real time. You can also perform some actions. Hang-up channels and Transfers via drag and drop. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. It's a work in progress... so

XLite does not support transfer... You have to buy their XPro -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mateo Meier Sent: Tuesday, February 22, 2005 3:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed to forwarda call to X-Lite.... Hey Guys Im

Try the snom soft phone! http://snom.com CS > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Dave Chase > Sent: Saturday, February 26, 2005 12:31 PM > To: ich@mateo.ch; Asterisk Users Mailing List - > Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Anybody using

PLEASE RESPOND IF THERE'S A SOLUTION I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the

Hi, I'm currently looking for a softphone for windows, we have been using X-Pro but it appears that X-pro doesn't support Message Waiting notification. Does anyone know of a well featured softphone that does support MWI ? I can't seem to find one. Any suggestions would be most appreciated, Thanks, Simon Ward

im using g729, but the bw usage is ~26 kbps per call, my gateways (cisco) support g723 and the bw between the gateways is ~18 kbps per call. Much better than the ~62 kbps of the g711. if you plan to be a voip provider you "must" go with compression codecs, especially if you want your customers to browse the internet while having a call. i.e. : We give voip phones (grandstream) to our

Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50

We have recently install an asterisk solution with about 60 physical extensions. While the system is running it runs reasonably well (Still have a few teething problems) but twice now they have experienced a degradation in voice quality and dropped calls and then finally asterisk completely crashes out. Restarting asterisk will work for a little while and it will crash again, each time less time