similar to: Asterisk + XEN does it make sense?

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 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, I'd like to use the convenience of apt packaging, but debian sarge's default asterisk is something like 1.0.7. Are there any apt repositories which provide newer versions of the software? Thanks! -- 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 List, Do you know if there are any plans to improve i18n for Asterisk? The current i18n way of doing it with asterisk is very limited and most of the time does not work. For example, take voicemail: "message" "received" "at" "seven" "30" "am" might sound good in English. But: "message" "recu" "a"

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, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi->exec ("DIAL $dialstring"); my $answeredtime = $agi->get_variable ("ANSWEREDTIME"); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the

Hi List, I know that the zaptel modules have echo cancellation, but is this possible to do this on VoIP <-> VoIP traffic as well? I'm toying with a SIP gateway which has apparently a terrible call quality and would like to know if there is any way asterisk can help with this. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP

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:

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

IMHO! I just see a skype channel as something good for asterisk. Skype has broad coverage. I can't imagine that skype wouldn't be interested in selling corporate accounts "skype trunk lines". Imagine having unlimited or X amount of continious calls coming in on SkypeIN and out on SkypeOUT from Asterisk. Internal Phones would all talk IAX or SIP to asterisk and use all PBX

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

Hi List, After quite a bit of struggle, it looks like I'm all ready to roll out prepaid cards on my small island. I now have a 4 E1s with a bit of spare capacity in order to accept incoming calls, and I can route Reunion Island mobile and fix through my own installations. For all other destinations, I need a carrier. I need good wholesale prices to Comoros, Mauritius, Madagascar, India,

Hi List, I would like to know if there is a way to detect gateways which time out (because of network problems or hardware failure for instance) when you send traffic to them. So when you do: Dial(SIP/number@gateway) If a call couldn't get through because the gateway has timed out, i want to do something about it. The idea would be to suspend gateway which time out for 60 minutes,

Does anyone have a working connection to Vonage via asterisk? (SIP, not ATA) I just signed up to test their service and they sent me a Number, Proxy, port and password. Every reference I have tried leaves me with a 404 error coming from Vonage. If you have a working setup, please post some config references. ? Thank You, Steven BerkHolz Soon to be known as HIROTEC AMERICA

Hi, I've had a bit of a problem with one way audio, and it happens exactly when I believe it shouldn't (and works perfectly when I would guess I could have issues. Setup: GrandStream GXP2000-------Linksys Router-----------Internet------Asterisk box (hosted somewhere, fixed IP, no NAT) ----------- VoIP provider -------PSTN When a call comes in from the PSTN, the call goes all the way

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.

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