similar to: IAX2 Problems causing server to hang

Hi We've recently set up and are using with success 1.0.7 using a Junghanns quadbri card to BRI ISDN, and Firefly with IAX2 protocol as softphones Works very well, however we're getting cases where sometimes the call just drops. >From setting more verbose modes we get a log which is shown below. The problem seems to be the maxretries message which comes from chan_iax2. We are using

Hi all, I have here several IAX2 Softphones(IDEFISK, DIAX and an own develop based on iaxclient.lib). I have follow dialrules in my std-test extension: [std-test] exten => *601,1,Answer exten => *601,n,Dial(IAX2/pbxnetwork/xxxxxx,30,m) exten => *601,n,Hangup exten => *602,1,Answer exten => *602,n,Dial(IAX2/pbxnetwork/xxxxxx,30) exten => *602,n,Hangup No I have a problem when

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

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 everyone, IAX2 softphone is not working with the “Unregistered” part in the asterisk (chan_iax2.c and data_pgsql.c) But with the Xlite softphone the unregistered worked properly and ast_data properly updated the IP address and port number in the database. I have seen some codes in the chan_iax2.c file: “ast_verbose(VERBOSE_PREFIX_3 "Unregistered '%s' (%s)\n", p->name,

Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

All, I've almost got my Asterisk PBX setup, but I've having some problems with the VoicePulse IAX trunk. On outbound calls, when dialing a PSTN number through the IAX2 trunk, music on hold (moh, using the m option in the dial command) does not work. The console states that "stop sound" on IAX2 channel. Ring works, but only without the r option. MOH works when trying to dial a

Hello All, I have been looking for a commercial version of an IAX2 Softphone for Windows but the ones I have came across (i.e. Iaxcomm, Iaxphone, Diax) do not seem to have an updated version since April 2004 in some cases. We looked at Firefly but we sent emails to Virbiage/Freshtel with questions and could never get a response from them. Has anyone got any recommendations for commercial

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

I posted earlier in the day about needed help with IAX trunking. I did some more reading and made some more changes. Here is what I have thus far: Iax.conf on one server [general] bindport = 4569 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw allow=gsm mailboxdetail=yes [vvfarm] type=friend username=colo secret=testpassword auth=plaintext host=64.194.211.170 context=iax-incoming

We just released a new version of the idefisk iax2 softphone, version 1.21 beta, available for download at http://www.asteriskguru.com/tools/idefisk_beta.php Some bugs were fixed, some new bugs might have been introduced :) - The problem with delays is finally gone!!! (one of the bugs was a memory leak, everybody using an older version is encouraged to upgrade.) Privacy Warning: Version 1.21 of

All, I have a simple question and a complicated reason for asking: Is it possible to change asterisk's source port for outbound IAX2 connections? I've tried using "sourceaddress" to no avail. I can set it to: proper.ip.of.box:4569 or 0.0.0.0:4569 and it works as expected. But if I try to set it to: proper.ip.of.box:5000 or 0.0.0.0:5000 it fails around line 8536 in

I have an inbound and outbound account with Voicepulse (I am very happy with the service, btw). But I have a weird IAX2 problem. When I get a inbound call on my Voicepulse DID, the call hits my asterisk server correctly with the correct callerid (the DID phone number 617902xxxx). when the call gets passed on to a softphone (X-lite), the caller id that shows up on the X-lite softphone as Lee ,

Hello! I'm looking for TEXT sending over IAX2, more precisely sending text messages between IAX2 softphone and asterisk. Currently I'm trying iaxComm and I was able to send a text message from asterisk to iaxComm (with AGI SEND TEXT). After some searching I got some questions: 1. is it possible to send text messages during a call (established or during ringing)? 2. is it possible to send

Hi, which OSX softphone do you use that supports IAX2 protocol with Asterisk? thanks Mike

> 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

Hey everyone here's my problem. Have a queue configured, it plays the desired recording, checks to see if agents are logged in via agentcallback, forwards the call according to distribution method, times out according to timeout settings, logs out the agent that did not answer, hunts for next agent, logs the rest of the agents out one by one when they don't answer, and drops call into

Hi, I have two asterisk machines, chomper and otao. otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no PSTN connections. chomper is at my house, behind NAT, but has a single X100P PSTN connection. I would like to establish two way calling between otao and chomper. Right now, I can call my extension on otao (2101) from my x-lite softphone on chomper, but I cannot call

Hi, I have two asterisk machines, chomper and otao. otao is otao.ieworks.net, has a public IP address (66.101.11.61), but no PSTN connections. chomper is at my house, connected to cable modem behind NAT, but has a single X100P PSTN connection. I would like to establish two way calling between otao and chomper. Right now, I can call my extension on otao (2101) from my x-lite softphone on

hello, please any IAX2 ActiveX control that wrap libiax2 or libiaxclient? i want to develope my softphone in delphi thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com