similar to: IAX2 Softphone Quality & Network Cards

> Hi! > >> We are in the process of an Asterisk call center deployment using IAX2 >> G711 ulaw softphones. Outbound sound quality is terrible. RESPONSE: This is Bill McCready from PCPhoneline.com . To address your sound quality issues, you may want to check out our VTA1000 Skype+SIP+IAX2 Tri-Mode Gateway which plugs into the USB port of a Windows computer. You can use it

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls.

Hi all, We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do.... Thanks....

Hi List; I need to configure a softphone to be client and use it with Asterisk, which is the recommended one? Is it iax2? Regards Bilal ____________________________________________________________________________________ Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games.

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a softphone for windows that supports IAX? I can't seem to find anything out there...maybe im looking in the wrong places... Jeromy Grimmett VoipEmpire.com jeromy@voipempire.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/e668136a/attachment.htm

I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here:

I am switching from IAX2 to SIP for my inter-Asterisk transport due to assorted quality issues following the 1.2.4 upgrade. On the server that SENDS the call, I have the following in SIP.CONF: [192.168.1.2_OB] type=peer fromuser=OB host=192.168.1.2 And in EXTENSIONS.CONF exten => 91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@192.168.1.2_OB) On the RECEIVING Server in SIP.CONF: [OB] type=user

I am attempting to update my Asterisk installation from 1.0 to the latest stable version. When I use CVS checkout, I am receiving the following messages on chan_sip.c: RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.510.2.25 retrieving revision 1.510.2.27 Merging differences between 1.510.2.25 and 1.510.2.27 into chan_sip.c M asterisk/channels/chan_sip.c Then, when

I just set up a Polycom 500 on *. Every few calls I make, the call connects and the receiving party can hear me (thru Broadvoice), but I still get ringing on my end, as if they never picked up. * logs look just fine. Does any one have any suggestions? Thanks. ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

I have call center PCs that switch between an IBEAM SIP softphone and a NEBU IAX softphone (for reasons that aren't germane here). The SIP softphones work fine, but the IAX softphones get a fast busy unless I give them an IAX trunk to use, instead of the PRI trunks that all the other phones are using. I am using Asterisk 1.2.3. svn rev 47264. I've appended a sample call trace. The

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On Tuesday 21 March 2006 11:19, Adam Robins wrote: > All switches and routers give

Ok this is my first post and I will try to keep it short. I have searched everywhere and haven't found an answer to my question I have two Trixbox servers that are connected over the Internet via an IAX2 connection. We are experiencing very poor sound quality. I have tried many different codecs gsm, ilbc, g729, g711 and all seem to have the same problem. (All though g729 seems to work the

All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4

For all the peoples that wanted to test my windows IAX2 phone, I've put it up on a server where it can be downloaded. For the ones that wanted the DLL, it's available on the same page. For the DLL, I will post a list of the functions in it and the parameters it expect as soon as I have some free time. All comments (good or bad) are welcome The phone can be used mostly with the keyboard

Can anyone out there please tell me what ports Speex uses? I want to set up QoS on switches but I can't seem to find this information anywhere. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended

I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569