similar to: OT: FWD Iax

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

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per

I am using IAX soft clients (firefly, IAXComm, IAXPhone) from a Win2K machine on a NATed private LAN to connect to my IAXTel account. All of my calls to outside numbers(1-800, 1-877, etc) seem to be disconnected after 9 seconds. Has this limit been placed on IAXTel? My call to the echo test number at 1-700-999-9613 seemed to run for as long as I wanted. --

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X systems. Sources included in the iaxclient library: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Precompiled binaries at: http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz Features: * Register with multiple servers (ie enterprise server and iaxtel).

Hi Folks, Just wondering if anyone can give me some pointers, I'm configuring Asterisk to talk to FWD's new IAX service. The asterisk server is behind an iptables NAT Firewall, with port 5036 forwarded: $IPTABLES -t nat -A PREROUTING -p udp -d $EXTERNAL_IP --dport 5036 -j DNAT --to-destination 172.16.20.200:5036 I can make outgoing calls just fine, but when I receive an inbound call

Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. In iax.conf, I have something like the following: [General]

I have set up FWD via IAX service. I have tested the IAX service with 613, echo test, and 612, saytime. It all works well. However when ringing a FWD user, I got this error all the time: Connected to Asterisk CVS-v1-0-02/01/05-09:34:45 currently running on chat (pid = 8282) chat*CLI> Verbosity is at least 3 -- Executing SetCallerID("SIP/1001-a1fb", ""David

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX<->PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

I'm attempting to get two Asterisk 0.4.0 PBXes to communicate with one another using IAX/IAX2 trunks. I've managed to get a semi-functional NAT Firewall working as a PBX (with Asterisk running directly on the firewall itself), but there are issues with bind()ing to various interfaces which is causing outbound SIP issues. To get around these issues, the idea is to do something like

Hi, I've just compiled and installed Asterisk (1.0.5). After some problems with codecs I could successfully connect to server by: guest@192.168.0.1 Next I created account at iaxtel.com and configured iaxcomm to work with this account. Unfortunately after that I had problem with logging as guest. Calling as guest@192.168.0.1 tried to connect with my local server through. So I changed it

I updated from CVS yesterday and today and still have the problem. IaxComm cannot transfer the call when it's an outgoing call. ('outgoing' is from the dial plan point of view). details : First I call the IaxComm phone and accept the call. Then I'm not able to transfer it from the IaxComm phone. If the call is an incoming call it works fine. details : First I call a phone

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

I found this configuration file on Wiki for FWD behind firewall ; SIP Configuration for Asterisk ; [general] disallow=all allow=ulaw port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to externip=xxx.xxx.xxx.xxx localnet=172.16.1.0 localmask=255.255.255.0 context=inbound-sip ; Default context for incoming calls maxexpirey=180 defaultexpirey=160 tos=reliability

Talking to myself ... ;-) Solved this by ... disallow=all allow=gsm ;allow=ulaw ;allow=alaw -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paulo Mannheimer Sent: quinta-feira, 11 de dezembro de 2003 09:02 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Iax, Iax2 and Iaxcomm Hi, I'm trying

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]:

Hello all, Can someone send me a current working example of what the entries need to look like for iaxtel and the iax.conf file inclusive of the register line? It's been quite awhile since I have done anything with iaxtel and I need to do a little bit of testing. Currently it appears that I can make calls out to iaxtel fine however inbound seems to be another question. I see 2 iaxtel

I launched www.goiax.com last week, which is intended to promote the use of IAX as a free and open source alternative to products like skype. There is no charge for the service. Right now I have free outbound to united states toll-free and us domestic numbers working. Currently the site hands out a virtual 87820-xxxxxxx number but I intend to add the ability to get a free United States DID

[ Sorry, I incorrectly copied some Reference headers into this post and tacked it onto the wrong thread. -Steve ] So far, I have been able to receive incoming iaxtel calls via my assigned 1-700-xxx-xxxx number, but only when using md5 authentication in iax.conf: [iaxtel] type=user ; Incoming calls only context=incoming auth=md5 secret=<mysecret> ; Required for