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

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, 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

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, 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

> -----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'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

Would anybody have any numbers on how large a box would be required to convert 100 or 200 SIP calls to IAX2, without transcoding, echo cancel, .. Or a setup with individual IAX2 calls coming on one side, and trunking being used to 1 or more remote boxes on the other side, to improve bandwidth usage ? It doesn't matter if you don't have a test done for exactly 100 or 200 calls, I'm

Hi there, I've asked this question to the BSD group too, but I'd like to know whether anybody else had similar experiences on Linux 2.6.20 etc.?? FreeBSD 6.2 Asterisk 1.4.5 (and 1.4.3 from ports) Sip phone - PBX(*) -IAX2-VROUTER(*)- SIP-Voip provider (SPA901 & SPA922 phones) We've see a situation where the IAX2 appears to "loose"/drop the voice data to be sent to

Hi, My question is in refernece to the posting by Jeremy McNamara here.. http://lists.digium.com/pipermail/asterisk-users/2003-October/022966.html He states that in order for "trunking" to work the type has to be peer.. When I set mine up I did so using type=friend just to make it simple.. So when I read the above posting I thought well maybe my "trunking" has not been

I wont say we are extremely high volume (40 concurrent calls) but I get occasional complaints about quality. Setup (at same location): Asterisk 1.4.26.2 FrontEnd Asterisk 1.4.26.2 Gateway with Sangoma A108D card with 2 PRI and LDT1 Connected via IAX2 trunking on its own VLAN Is IAX2 the way to go or would SIP trunking be better. I know its a pretty vague question but I am just trying to

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 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

Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks, Geo ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

Im having problems setting up a trunk between two locations. Heres the setup I have: Server A is connected to the PSTN at my datacenter Server B is connected to a clients e1 line at his datacenter I only want to route calls from Server B to Server A and out through the PSTN. Server A has a lot of other things connecting to it, so I need a very specific context for all calls to go through.

Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive

I am curious if anyone has had problems trunking iax2 with 100+ concurrent calls. I am planning on testing this out tomorrow, however I wanted to know if anyone else has had a problem with this prior to my test so I know what to look for if anything is known and what resolutions have been found if there are any known problems. Specifically I am doing this on fbsd 6 with asterisk 1.2.4 using