similar to: iaxclient softphone: quality?

Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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.

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

Hi All; If someone used speex and has experience with its settings, then who can help to explain the following: 1) When it is recommended to use VBR (vbr => true)? 2) If there relation between setting the vbr => true and the abr value (for example to be 0 or 1 or 10) and the relation between this value and abr (true / false). 3) Any relation between the quality value and the abr value?

Hi, I've had a hard time understanding what was going on in a new * setup. The deployment has a * box running on dual xeon RH9 stock 2.4.20-8 and some different versions of asterisk (1.2.10/1.2.16) + libpri + zaptel + wanpipe. Short version: audio from iaxclient clients is fine from windows but poor from linux when going to zaptel. E.g. Iaxcomm running on windows works fine, but the same

Hi All; Is there a softphone that can be installed on a mobile (new mobile models), so it can work with Asterisk as following: 1) As SIP or H323 client, with the ability to add button functionalities (call pickup, call transfer, ...) so if there is a wireless network, then it can use it to connect to Asterisk and work as client, but from the Mobile. 2) If there is no wireless network, then it

Hi all; I have two asterisk boxes installed in two separated sites, the internet bandwidth between them is very good and I am using G729 codec to communicate with them and IAX. The problem that side A hears well side B while side B does not hear well side A !! I did one thing in side B that in iax.conf, I set the bandwidth=high and it helped, but still side B is complaining from the quality

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over

Hi All; I discover that most of the voice cutting complain are coming from the Internet bandwidth when we are connecting two remote offices togethor via Asterisk or any other IP PBX. Anyone has an idea on a ADSL router that work as ADSL + Bandwidth division? So we can resolve the problem of providing a guaranteed bandwidth for the voice packets instead of suffering the voice cutting? Regards

David Brazier wrote: > Hi > > The latest SVN trunk for speex has changed the SpeexPreprocessState to > an opaque structure, for jolly good software engineering reasons. > However, the Analogue AGC (AAGC) feature of iaxclient (in audio_enode.c) > relies on some members of this. It uses speech_prob to detect when > there is enough speech to consider AAGC and then loudness2 to

Asterisk-people, Some of you may have heard that we were working on a simple, cross-platform IAX client library called "iaxclient". We've pretty much been "on vacation" with the project for a while, but recently have made some progress, and now have the library working across platforms, and a simple test client called "testcall", up and running on 3

Hi All; Can I run the IAX on TCP port instead of UDP port? If I ran IAX in TCP port, and in case my network was having a lot of users doing browse on the internet and downloading, so in that case and if the IAX used TCP port, so the voice will be better than using UDP (because in TCP the lost packets will be resend while in TCP it will not which will cause the voice to be cutting)? Same thing

Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Regards Bilal

Hello Folks, I noticed that all of the IaxClient based softphones with exception of Firefly only seem to have support for GSM but what about iLBC? The quality is excellent with iLBC even on a dialup connection! Meanwhile while the audio on GSM often sounds scratchy. Is anyone looking to implement iLBC in an IaxClient based softphone soon? Errol Biz4Web Solutions Limited

Hello; I have a SIP trunk with a service provider, the caller from landline or mobile is hearing us very well, but the agent who is sitting on the handset is not hearing well, the voice at the agent is not crystal (like he is talking from well or far deep place). Although the IP Phones are cisco 7942G and the used codec is g711ulaw (actually it gave better quality than g711alaw). If we increase

Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

On Thu, 08 Apr 2004 10:14:09 -0400, Steve Kann wrote: >Gary wrote: > >>I have noticed lack of codec negotiation with calls thru a registrated >>asterisk box. >> >>No seen problems with outbound calls, (though I haven't specifically >>tried it), but the problem exists inbound. >> >>Easiest method for testing this was ring in via a sip client set

Hello; What is the best method to let the voice quality through Dahdi channels to be clear and no echo? Is it the wanpipe or it is working only with sangoma? Regards Bilal

Dears; I need to order Digium card and not able to know which one is the best quality? Is it that of AEX with the end E or EF or P or B? I saw those card that its slot is small (I think those that end by EF), are they the best card? Really I am caring to have a card that has echo cancelation and the voice volume is high enough (because previously I faced the problem that the voice volume was