similar to: Change g729 payload

Hi all, Eyebeam has a sip-chat function and it would be nice if I would be able to use it. But the problem is that I can't really find information about it. I can just try to send a message and on the Asterisk console a message like this appears: Jun 13 10:05:25 WARNING[6512]: chan_sip.c:7281 receive_message: Received message to <sip:bla@voiphost> from "Bla

Hi all, I'm trying to make a context that will monitor a call and when it's completed it would e-mail the wav to a specified mail adres. So I made a standard context that records a call, like this: exten => _*31*00[1-9].,1,Setvar(CALLFILENAME=CALL-${EXTEN:4}-$ {TIMESTAMP}) exten => _*31*00[1-9].,2,Monitor(wav,${CALLFILENAME},m}) exten =>

Hi All, First off all, this is my first mail to this mailing-list, so if I am doing something wrong please tell me. And apologies for my english in advance, it's not my native language. Anyway, I have few machines running Asterisk 1.2.7.1. All machines but one are Gentoo (other one is Debian). The problem is that Asterisk keeps eating my memory. Just random (mostly at night) all my free

Hi All, I need a function that I believe isn't available in Asterisk, but I don't know if I'm correct about this. I have a queue and I want agents that are in that queue to have the ability to answer a call in the queue with calling an extention. For example, if I'm an agent and my colleague forgot to logout I could take the call when his phone is still ringing without

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, 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

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 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, 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.

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

Hi, understand that the bandwidth utilized for each call is dependent on the codec used, wonder if Asterisk can monitor the total bandwidth utilized and restrict/reject new calls when the resource is insufficient to support them reliably? Regards Andy Tan -- Andy Tan andytan@fastmail.fm -- http://www.fastmail.fm - Does exactly what it says on the tin

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 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

Hello all, Just a question, on asterisk box : I looking on the web , for asterisk at large , and 'asterisk future of telephonie' ... If we would like to change our OLD PABX 600 phone with 4 E1, to install a asterisk with full ip phone in SIP, Could we use 1 Box for asterisk with voicemail, zap channels and some agi script ? thanks Fabrice

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.

You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN

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

Hi, Does anyone know what is the amount of max concurrent calls that can be made in one Asterisk box? I heard that it is 256 and it doesn't depend on how good your machine is. It is the program constraint. What can I do when I need to have more calls than that. I read about connecting Asterisk boxes with IAX. Is it a good solution? Does anyone have other proposals? Cheers Andrew

Need help...I need to install a card to terminate 7 lines...I have not order the phone lines yet...I can either do analog lines 1FBs or order a fractional T1...any suggestions on what hardware would be easier to install and configure...also if I went with a T1...do I need an external CSU/DSU or anything or does it just plug into the T1 card...thanks.. -----Original Message----- From: