similar to: SIP & IAX behind NAT

Hello! I have a number of IAX clients behind a NAT (on the same LAN) and asterisk server on the Internet. And that clients doesn't speak directly to each other, traffic goes through the asterisk server. What should I configure to make IAX clients on the same LAN to speak directly, please? notraster=no is set in iax.conf The asterisk server is on real IP behind a NAT (at DMZ with full 1-to-1

Hi, I'm searching for GPLed softphones. I found WengoPhone but actually not available for Asterisk PBX, only for Wengo network. I found Kiax but only for IAX protocol. Did you know a good GPLed softphones which works on Windows ? Thanks Greg -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi together, I try to find a softphone (freeware) solution for Windows 32, that works without problems ... Right now I use iaxcomm wich was best, of the ones I tried. But I have one problem with it. When I turn on qualify, it will not connect to the asterisk. This is also documented and normal behaveor. But if I turn of qualify, iaxcomm does not reconnect to the server, when the server got

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

Hi, IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 second. The two phones were on 2 different pc's near me. When I speak on one, i hear it on the other after about 1 second. I tried using iaxComm, Xten Xlite, etc. Same. FYI: The codec used was GSM. Using the fxo and fxs interfaces on the digium cards with POTS have no such issues. Any clue where the

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.

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.

Hello I have created an iax exten in my iax.conf file: [300] type=friend username=300 secret=*** context=default host=dynamic callerid="some name" <300> auth=md5 Then in my extensions.conf I have: exten => 300,1,Dial(IAX/${EXTEN},20) exten => 300,2,Hangup I can dial from iaxComm (a soft IAX client) and that works fine. But when I try to dial 300 get: WARNING[22077]:

Hi, I have configured 3 users in my iax.conf, i am using iaxcomm phones. Iaxcomm has excellent voice quality although there is no ringing tones(either ring back or ringing tone),but i can live without right now. I find that for each user i want registered i have to add his name and his ip address.I have been using "host = dynamic".Isnt there any way that i can define a dialmap such as

Hi folks an issue I don't understand. I'm running * stable 1.0.3 on public internet, with following iax.conf / sip.conf entries: iax.conf [100] type=friend username=Foo context=default auth=md5,plaintext,rsa secret=secret host=dynamic callerid="Foo" <100> qualify=no sip.conf [10] type=friend username=Bar context=default callerid=Bar <10>

I wonder how IAX compares to SIP bandwidth-wise? I've tried both over overseas IP connection, and somehow SIP seemed to work better. Peter

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

Hi, I seem to be having problems with IAX clients based on the iaxClient library. I have been working on my own client (an augmentation to the Call Manager I released last week) and it seems to regularly miss incoming calls entirely. It also occasionally misses the drop signal when the remote end drops a call. Has anybody else seen this kind of behavior? I have tested with my client, with

Folks, It seems like the * v 0.9 and iaxcomm won't speak to each other. Is there another IAX2 client that is usable under Linux (Debian preferred)? Thanks, Tim --

--- Peter Zeltins <peter@fintrading.com> wrote: > > > Well, I happen to be one of those very specific cases... ;) and looks > like > will have experiment with it myself. Although I'd hate to re-invent > the > wheel. > > Peter Checking e-mail this morning it looks like we have two independent "fixes" that both do what has been suggested in this

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

Hi, I'm a student and my thesis work consist in testing Asterisk with Swyx(SwyxWare). My approach is to declare asterisk as h323 gateway for the Swyxserver using oh323 Plugin. Is there any possibility to connect Asterisk with Swyx? how? the outgoing call must pass from Swyxit->to Swyxserver-> Asterisk->to PTSN Thanks

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....

iaxComm is a crossplatform open source softphone utilizing the IAX2 protocol. It is distributed as part of Steve Kann's iaxclient library. I've just posted new Windows, Linux and Mac OSX binaries to sourceforge. The Windows binary was compiled on WinXP. The Linux binary was compiled on RedHat 9. The OSX binary was compiled by Andreas Wrede on 10.3 and was tested on 10.4 (Tiger) beta.