similar to: IAX2 + Jitter Buffer

Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing

Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI> Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas?

Hi List, I am working on least cost routing code on the moment, and I am stumbling on a problem. Say you have provider A having: Prefix XXX 0.10 Prefix XXXYYY 0.20 And provider B having Prefix XXX 0.15 You're stuck, because you cannot decide if provider B's "XXX" prefix also covers XXXYYY numbers or not. If it doesn't, it would be a waste

Two cvs-head asterisk boxes with iax2 working fine (without register statements). When two calls are placed simultanously from system A -> B and the packets are sniffed on the wire, I see the two calls using two different udp packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes (at both ends). I was expecting to see both calls handled within a single udp packet, but

Hi List, I am looking for a 1 FXO analog termination device, other than the obvious PC + FXO card, and which can achieve decent call quality. The SPA-3000 seems an option... have you got any other ideas? Cheers, Jean-Michel.

Hi List, Being very new to Xen I have a few generic questions for the list, I hope to grab some useful advice and pointers to documentation. I am evaluating Xen to consolidate a few existing servers into one appliance (mainly in order to reduce power consumption, heat, and hardware failure risks). I plan to have a SER router, an Asterisk LCR router, a voicemail server, a calling card server

Hi List, I've had some issues with some VoIP providers where either: 1 - There is massive PDD but finally the call goes through 2 - There is massive PDD but the call gets rejected anyways I was wondering if there was a way to automatically detect such behaviors when it happens (maybe with a script or something) so I can take the faulty providers out of the routing and maybe automatically

Around 1978, when I was consulting to a multinational company in the business of agriculture, I witnessed this configuration in their communications center in NYC: A paper tape punch attached to a teletype machine was busily punching out a tape that was being spewed into a wastebasket. Somehow, running behind it by several feet of tape, was a paper tape reader on another teletype drawing

Can anyone point me to a reference or sample config for bypassing a nailed up (point to point) t1 between two PBXs with asterisk and a pair of t1 cards? Right now I have 2 Nortel norstars connected to each other via a leased line t1. I also have a solid 10mbps low latency microwave link between the 2 sites. My goal is to run an asterisk box at each end with a t1 card and Ethernet card to

Hi List, Is there a way to tell asterisk to only accept SIP streams from the same IP address that is used for signaling? Thanks, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

> -----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 >

Hi List, I was wondering if anybody had tried running Asterisk inside virtualization software such as Xen. Are there known problems doing it? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality

RAD appear to have bucketloads of products which bridge between various interfaces (E1, BRI, POTS) and their own TDMoIP protocol. The attractive thing about them for me is their availability in Australia. The voip wiki says not much about it (http://www.voip-info.org/wiki/view/TDMoIP), and certainly nothing about if there is any way to get Asterisk to talk TDMoIP. Despite the name, TDMoIP

Hi, I need to have some SIP extentions on remote places where the latency from my asterisk box with public ip is 1~1.5 seconds. What codec will work fine on this sceneary? I'm planning to use iLBC, is a good choice? Regards, Guillermo.

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

We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I'm personally drooling over the UTStarcom F3000, but compatibility and shipping ETA info is a bit sketchy. Phil

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

I've just stumbled across a rather weird problem and was wondering if someone could shed some light on the situation. In testing faxing through Asterisk using Voicepulse Connect for trunking I am able to receive faxes without a hitch. Quite impressive considering previous experience with certain other VOIP providers. Today I finally got around to testing outbound faxing and found that if

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the