search for: softfone

...my computer The problem is this that i get my voice back mean there is too much echo(there is no complain from the caller). I have set following values in zapata.conf echocancellation=yes (also tried different powers of 2) echocancelwhenbridged=yes is there any other setting or not ??or this is a Softfone prob.??(i m using SJphone ) any help in this regard would b very helpful for me Obee _________________________________________________________________ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail

...happens is, when I do a transfer using the Transfer button, the phone, does not play the music on hold, which is waiting on the phone is silent, and I have the same settings on a 1.4 server, and the music plays correctly when using the same phone. When using the * 2 to transfer the connection or a softfone, the music plays correctly on this server with Asterisk 1.6. What the detail is missing in my configuration? My Configuration [featuremap] blindxfer=## atxfer=*2 automon=*1 disconnect=** I made a DEBUG to use the channel when the two key TRANSFER Server 1.4 and 1.6...

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys work fine calling a URA menu on the Asterisk and can talk to each other with excellent audio, only SIPURA->PSTN(HICOM) gets garbled....

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be

I have a call center where we're looking at converting it from a traditional PBX w/ digital phone "agent" sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we're currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad,

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To: asterisk-users@lists.digium.com Sent: Tuesday, March 21, 2006 14:21 Subject: SIP Realtime 1.2.5 and Username/auth name mismat...

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ