similar to: Hypothetical IAX2 situation

Hello all, I updated from CVS 3 days ago and now my IAX2 gateway is dropping calls without warning. It happens right in the middle of a conversation with no pattern. I never had this Problem before and am usually talking 2-3 hours a day. Is their a bug? Should I rollback? Cheers, Paul Seniuk -------------- next part -------------- A non-text attachment was scrubbed... Name: Paul

Unfortunatly no on both counts. The arrangement right now has: PSTN Trunks & Stations <-> Nortel Norstar#1 <-CT1-> Asterisk#1 <-IAX2-> Asterisk#2 <-CT1-> Nortel Nortstar#2 <-> Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there have been reports of PSTN calls

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

Or perhaps how to configure and refer to two parallel IAX trunks with different codecs? I have a situation where I'm using G.729A as my IAX trunking codec. Now I need to push some short duration, low bitrate modem traffic over the link (a credit card terminal). Obviously the modem audio isn't going to survive the G.729 codec process intact, so for the times the device is used I'd like

Hi all, I'm running Asterisk 1.0.0. I am a customer ( and supporter ) of voicepulse. For me, it works perfectly, but one of my customers noticed a small problem: During a conversation, when the otherside isn't talking, it's almost like the mic turns off. Not that big of a deal I know, and the more I think about it, the more this seems a voicepulse issue. But in the off

HI, I am trying to setup winbind on Samba 3.0.2 running on Red Hat AS 3.0. I have completed most of the steps of setting up winbind successfully but when it came for me to login in using the AD account username and password, it didn't allow me to login. the error message i am getting is incorrect password or check username. During the setup i tested the wbinfo -u command and i was

Hi Is anyone successfully using iax2 trunking with CVS head ? The reason I am asking is that I have heard there may be some audio problems, which I would like to know about before sending customer's calls over a iax2 trunked connection. Thanks in advance. Clive

Hi all, I've got an old avaya partner acs <7.0 system here. I'd like to add a simple voip bridge so I can hook up our remote offices. From my research, it would seem the pre-7.0 series doesn't have a t1 port, so if I wanted to do this, I would have to feed the avaya system fxs ports from the asterisk box. Does that sound about right? Has anybody ever done this? Does

Hi folks, I've got a few machines that I want to setup winbind on, so they can auth their users against the AD win2k server I have here. On top of that, I want to map their home directories from one to a few others. So, I'll need to make sure their user databases are in sync. No problem, I thought, I'll just nfs map /var/cache/samba and be done with it. Well, as it turns out,

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

Hi all, Before I setup an account with them, I'd like to hear other people's impression of LiveVoip. I'm considering using them for 800 numbers, and I'd like to feel comfortable that others here on the list have had good experiences with them. Thanks, sorry if this is the wrong list for this. :) Sena

On Thu, May 31, 2012 at 05:39:30PM +0200, Sander Eikelenboom wrote: > Hi Konrad, > > What is the status of the "devel/netback.liu.v5.on.3.5" branch in your tree ? Hm, lets wrap Ian and CC in this converstation. Were there any big outstanding TODOs before the network maintainer was OK with them? > It doesn''t seem to be destined for 3.5 mainline yet ? > >

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

I've got a need to do something like the following: [foo-context] exten => _.,1,SetCIDNum(123) exten => _.,2,SetCIDName(XYZ) include => local include => tollfree But of course, this example won't work. The goal here is this: if a call ends up being handled by the "local" or "tollfree" contexts, I want those SetCID*** commands executed. Otherwise, I

I am getting ready to experiment with the Tellabs 2752 echo canceller. I have a 255D shelf (and power supply), but am struggling a little on connecting the echo canceller to a PRI. The shelf has 4 25-pair amphenol connectors. The two on the line side are marked "Receive In" and "Send Out". The 2 connectors on the drop side are marked "Send In" and "Receive

Hello I have a Grandstream GXP2000 with latest firmware. When I use it holding the handpiece I don't hear any echo - neither does other end. However, if I use it handsfree, the other end notices echo when they speak - ie their voice is echoy. I hear their voice being a bit echoy. Is this purely down to the IP Phone? Is there anything I can do about it? I considered buying a more

I apologize for sending this to the list. Keith from Hazleton... your mail server is rejecting mail I'm sending you from my mail servers, as well as from gmail... you may really want to consider using a different blacklist.. the on you are using now is going to block almost everything and everyone.