similar to: IAX Huge Delays after Hold or Transfer

Hello, any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? Thanks for help. Mit freundlichen Gr??en / best regards Andr? Herrlich IT-Operator / Developer ____________________________ LetMeRepair LMR Service and Consulting GmbH Fichtestr. 1A 02625 Bautzen Tel.: + 49 - (0)3591 - 2722 - 1451 Fax: + 49 - (0)3591 - 2722 -

Need help on this.. I have configured all my internal extensions as SIP phones, i have one ATA-186 and one softphone. When I try to transfer calls between internalt, I use *1 as it configured on feature.conf. But my external line configured a IAX. When there is an incoming call from outside, the line 1 from my ATA186 rings. If I want to transfer my call using *1 is not possible. So I need to

In zapata.conf you can set a global default music on hold class for all Zap channels, or specify them on a channel basis with the musiconhold=class parameter. The same is true for sip.conf for SIP channels, except that the documentation lists the parameter as "musiconhold" while the sample configuration file comments list it as "musicclass". I have been unable to determine

I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read "Asterisk: The Future of Telephony" and done a fair amount of googling. I am completely sold on Asterisk, and the new building's

hi, I use IAX softphone with asterisk and I notice that a call between two IAX softphones end after 1 min. Then I can't hear anything but the call still in progress. I have this log in asterisk IAX debug: Rx-Frame Retry[No] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00016ms SCall: 21589 DCall: 00001 [192.168.1.22:4569] Tx-Frame Retry[000] --

Hi all, I am experiencing a problem with the quality of the voice communication between an IAX based softphone (WinIAX) and an outside line through a FXO port or even with a regular analog phone connected to a FXS port. The party using the IAX softphone hears his own echo a plit of a second after speaking. The party on the analog end does not experience any echo. I tried to modify the KFLAG

Hi, I have a problem with my IAX softphones. After a call, when the softphone hangup, it remains unavailable for the other softphones. It can call anybody, but can not be reached... For example, if A call B, B answer, then A or B hangup, and C won't be able to call A or B after that (but A or B would be able to call C). The Dial function returns that the chan is unavailable. That is very

I have installed several call centers in the netherlands with the eyebeam softphone (from the counterpath guys) It is not free, but very stable, and pretty easy to use. It works great with asterisk (specially the presence option, so agents can see whether somebody is actually ready to take a call). In combination with sennheiser headset CC series, I have had no complaints. We also use a tapi

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

>Why do people want to develop more softphones? There's already lots of softphone projects out there >that could use a hand, if you want to work on one. Starting from scratch will just add one more half >baked softphone to the growing list of unuseful Open Source applications. >Especially since you consider using an ActiveX control :) Actually, I have an application for a IAX

After having been toying around with asterisk and various VoIP stuff for a couple of weeks now, I want to recommend a preferred protocol and softphone to friends and family for calling me up. As SIP and H323 are such a mess to set up in NATed environments, the only reasonable protocol option right now seems to be IAX. After looking at http://www.voip-info.org/wiki-Asterisk+IAX+clients and trying

Hi guys, Does anyone know of a way where I can bring a third person in on my conversation. Say I'm on a IAX or SIP call from a softphone DIAX or IAXCOMM and am speaking to someone now I want to quickly bring another SIP or IAX extension into this call so the three of us can speak to each other. I know I could do this by transfering the first person into a meetme then calling the second

I'm using Asterisk to do audio as well as H.263 video over SIP. Actually the video works pretty well but I have trouble with the audio. I'm wondering if someone can suggest codec/jitter settings or other tweaks. The system looks like this: Softphone <---ulaw----> Asterisk #1 <------IAX (usually GSM)---------> Asterisk #2 <-------- IAX (usually GSM) -------->

Hi Yall, I would love to put a very compact phone on my wife's desk at work... Ideally this would be a very small IAX phone with 2 RJ-45's so I could drop it in without much notice and only have to beg for 1 port from the sysadmin. I have looked around and I don't see such an item in existence? What I would love is a "home" phone looking handset (wired is fine), which

hello List i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip

Hi all, We have an asterisk server running on redhat 9, with a TE410P card connected to 4 PRIs, on a dual 2.6GHz xeon server with 2GB of RAM. We also have 10-12 agents using IAX based softphones on winxp. Almost all of the usage consists of outbound calls done automatically through the PRIs and connected to an agent once answered. The agents perform the 'ready' operation via IAX by

Hi all, I'm having some trouble. I've got an IVR with 4 different languages using the SETLANGUAGE command and I'm getting a 6 second delay when I make the first selection, after that all is fine. There's nothing in my dial plan that I can find that would be causing it. And the delay is driving me nuts! I have an IAX connection from a provider coming in. Could this be the cause?

Hi, I am Chandramouli from India. I have successfully implemented Intercom (Dialling within my office LAN) and Voicemail features using Asterisk. To implement this, I am using X-Lite Softphone. Now, I wish to make calls to US using VoIP Asterisk. I dont think I need any external hardware to implement pure VoIP solution. Here I am sending my configuration file values: Contents of

Hi, I am not sure if it's a suitable place to ask this question. Anyone here familiar with USB phone please? I want to buy a cheap one to make it work with IAX softphone. Regards. Steven louse _________________________________________________________________ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail

Is there any winblows softphone available offering g729 *and* IAX? I couldn't find any http://www.voip-info.org/wiki-VOIP+Phones The best choice should be dIAX, but it is only GSM.