similar to: RC1 problem? (Conversation over two IAX2 streams = nasty, gappy audio)

Hi, With reference to Andrew Kohlsmith's problem with calls over IAX2 going dead after 1minute 5 seconds: This is caused by a bug in the "optimized bridging" code in chan_iax2.c interacting badly with the IAX2 jitter buffer. This problem only affect calls where: 1) There are 2 or more servers doing optimised bridging between the end-points of the call. ie a call like A-end

I just upgraded to RC1 from a two-three month old CVS , and noticed that during IAX2 calls to my service provider there are periods (usually less than 10 seconds long, minutes apart) during which the caller can not hear me, but I can hear the caller fine. Inter-office calls (SIP-to-SIP) does not appear to have this issue. Has any other users experienced this? Marcus Adolfsson TreoCentral

Well this is disappointing. Time to find somebody else... -- Wes -----Original Message----- From: NuFone Operations [mailto:support@nufone.net] Sent: Tuesday, April 18, 2006 3:44 PM To: wbaehr@totalmac.net Subject: NuFone Update: DIDs Effective 3pm EST Today, April 18th, 2006 Telesthetic, the carrier supporting the Toll-Free and Michigan DID operations of NuFone, has threatened to terminate

*1 and *2 are identical machines (single Xeon with HT disabled), one with a TE405P and the other with T100P. If I place an IAX2 call between them when both have trunk=no in their respective iax.conf sections, calls are fine. If I place an IAX2 call between them when both have trunk=yes in their respective iax.conf sections, I get "gappy" audio. What I mean by this: trunk=no:

Hello, I've signed up for a NuFone account, and added the following instructions to my config files per NufFones directinos: iax.conf [NuFone] type=peer host=switch-1.nufone.net secret=password extensions.conf (under the [default] context) exten => _1NXXNXXXXXX,1,Dial,IAX2/f00b3r@NuFone/${EXTEN} I then get this message in the CLI: -- Executing Dial("SIP/jake-fe5d",

We have a customer who is connected to our PSTN gateway using IAX and noticing that even when the traffic from their site is modest their outbound audio has short dropouts. Inbound audio is fine. (They have ADSL so it is expected that outbound audio would be the first to experience problems.) We have several questions to pose to the collective wisdom of this list. Q1: Are there any statistics

Anyone know how many simultaneous calls you can receive on a NuFone DID? -Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050310/e20d2eb8/attachment.htm

Anybody here having any problems with nufone? Calls are not going thru, when trying to call their customer service number it doesn't go thru. When trying to resolve www.nufone.net I get (sourec: http://www.dnsstuff.com/tools/lookup.ch?name=nufone.net&type=A ): How I am searching: Searching for nufone.net A record at j.root-servers.net [192.58.128.30]: Got referral to M.GTLD-SERVERS.net.

I am having a problem with your my nufone service. I'm trying to setup incoming calls and I'm having no success. Outgoing works fine though. The message I'm getting is "the person you are call is not currently reachable". I'm going to give you as much info as I can. I'm also an asterisk newb! Anyways, I installed asterisk@home. Set up extensions which communicate

Hi, I've been using nufone DIDs for months with no problem. Now there are codec problems that prevent any kind of calls working. For example, May 5 13:04:12 WARNING[928]: channel.c:2115 ast_channel_make_compatible: No path to translate from IAX2/NuFone@NuFone/25(256) to SIP/wengo-out-968a(4) May 5 13:04:12 WARNING[928]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't

Hello, I am trying to configure the my asterisk box here with the following **iax.conf*** [NuFone] type=peer host=switch-1.nufone.net secret=xxxxxx ***extensions.conf:*** exten => _1NXXNXXXXXX,1,Dial,IAX2/xxxxxxx@NuFone/${EXTEN} exten => _011N.,1,Dial,IAX2/xxxxxx@NuFone/${EXTEN} I have a couple of Xlite softphones and 2 analogue phones connected to a mediatrix 1102 connected to our lan.

Good evening, I just signed up with Nufone and I am able to receive calls with no problem via my 800 number. Outgoing calls are not going through though. My extensions.conf is as follows: [nufone-out] exten => _91NXXNXXXXXX,1,SetCallerID(mynumber) exten => _91NXXNXXXXXX,2,Dial(IAX2/user:pass@switch-2.nufone.net/${EXTEN:1}) exten => _91NXXNXXXXXX,3,Congestion Whenever I try to

I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called me@NuFone/1515480XXXX -- Call accepted by

Evening, I've got * asterisk up and running with nufone account for inbound and outbound calls from the world... everything works quite nicely except that I noticed last night that when I have a call active from * to nufone, web surfing on the network slows to a crawl... additionally, I have a VPN tunnel to another office (using CIPE) and while inter-office (i.e. - SIP to SIP) calls

Folks, I've ordered a new account from Nufone last month. Transferred money to Nufone through their paypal account. I had communication with Nufone sales up until two weeks back. Since then there were no replies to my emails. I am afraid with this kind of unresponsiveness how one would run a reliable service with this company. Have no bad feeling with Jeremy as the author of widely used h323

I am experiencing a very wired problem. Some of my cdr are lost. I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning. I am running asterisk 1.0.7; this is simple configuration file: extensions.conf [general] static=yes writeprotect=no [macro-gw-voipjet] exten =>

Is nufone having problems taking gsm calls today i had some issues dialing overseas to call my folks. here's snip of what the console displayed -- Executing Dial("SIP/2204-a279", "IAX2/fgravato@nufone/011351217907000|100|T") in new stack Jan 18 10:30:02 WARNING[1200884528]: chan_iax2.c:5036 iax2_request: Unable to create translator path for UNKN to GSM on IAX2[NuFone]/1

Ernest, I hadn't thought of doing that, though having that added protection would be nice. However, what I'm trying to do it have an incoming call at my home number follow me to my cell phone for selected numbers -- Since I already have three way calling, I'd like get Asterisk to essentially three way my cell phone into the call (or my office number, etc.) I understand the

Hello: . I'm having troubles registering on nufone's IAX service . I'm really new to Asterisk Nufone provide me some config examples ... I can dialout but I can't register my * Box, eg. whe I do "iax show registry" I got only a "Request Sent" and later I have a "Timeout" My box is on a Public IP and no firewall. I'm using *0.5 on a

Hi, everyone ! Looking at this explanation : "When SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. Once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the