similar to: Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled?

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

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

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk

At the moment we're not - the email notification from Comedian Mail has been mostly sufficient. I do however have some Dialogic D/42-NS PBX emulation cards and the plan is to use them to set and unset the MWI lamps based on events pushed out of Asterisk. They may be obsolete hardware but they came in real handy for extracting the voicemail from the old StarTalk NAM too. Take a look at the

Hi, I thought I'd repost this to the -users list for some background on the jitter buffer and its workings and remaining issue.s I'll also pu a little executive summary here at the top: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) The jitter

Couldn't see the forrest for all the fascinating tree-like applications that are out there: For future reference, see: http://www.voip-info.org/wiki-Asterisk+call+parking :-) -----Original Message----- From: Kris Boutilier [mailto:Kris.Boutilier@scrd.bc.ca] Sent: August 11, 2004 1:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Inband announcement of parking slot from

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

I am using a SIP softphone (X-lite, SIPPS or Firefly) connected to an Asterisk v 1.0.3 PBX. The PBX is also connected via a ISDN-PRI crossover cable to a Avaya Definity Generic 3 PBX via a TE405P card. All outside of the office calls go through the Definity. Here's the issue: Calls to internal SIP extensions, Definity extensions, other offices within our private network (through the

> -----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, Has anyone had success using a TE410P card in an HP-Compaq DL380 G4 server? For me the card is detected fine, but the system just never sees an interrupt from the card. I've tried everything I can think of. The card definitely works. Its Fedora Core, but we also tried a stock 2.6.10 kernel. We tried with and without Hyperthreading, with "noapic", we disabled all the

Consider this dialplan fragment, where the call is being dialed into [macro-process-routing] over an iax2 channel from another (same build) Asterisk server: [macro-process-routing] ; This is the entrypoint of the debug call but is also refered to by Macro(process-routing) elsewhere in the dialplan ; XXX-NNN-6800 exten => _6800,1,Macro(6800-interceptor) ; This is matched when 8 is

I built the CVS version of the Speex library - v1.2 it calls itself. Asterisk seg faults trying to use codec_speex.so. I'll have a look to try to fix it, but thought I'd just ask if anyone else knows what needs to be done? Steve

Inquiring (management) minds want to know. I'm assuming it's because 'it's funny how simple it really is to write a really decent voicemail system'? Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Gary Reuter > Sent: Thursday, August 11, 2005 12:59 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped > betweenpstn & norstar > > > I poured over my logs most of

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

Hi, I'm stuck on a silly thing. I need to get the "billsec" CDR value after a call. But I'm finding its always 0. Here's my test code: exten => *244*,1,Dial(Local/test@custom-tests/n,,g) exten => *244*,n,Noop(after dial duration is ${CDR(duration)} billsec is ${CDR(billsec)}) exten => *244*,n,Hangup [custom-tests] exten => test,1,Answer exten =>

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

Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while

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