similar to: Integrics release Enswitch 2.0

Integrics is pleased to announce the release of ITSP version 1.6. This version has the following new features: - Comes in 2 editions: * Carrier edition, for 250 to tens of thousands of users on hosted systems. Integrics sells this edition directly and through partners. * Office edition, for 10 to 250 users. This edition is sold only through our partners, for them to sell as PBX systems at

Hi, I have a growing voip business Im i looking a solution that can handle at least 3000-4000 concurrent calls with great performance. Also with a billing platform, reports, reseller platform, LCR, call routing,real time reports, SQL dababase access real time Load Reports. Any recommendation? Thanks Ignacio -------------- next part -------------- An HTML attachment was scrubbed... URL:

This is a request to the website manager for asterisk.org. The build scripts for our ITSP product include the URLs to download the Asterisk files, such as: wget "http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz" However, if a new version is released, asterisk-1.2.5.tar.gz is moved to the "old" directory. This breaks our scripts until we can update them and send

Hello, Can you recommend a good billing software for asterisk that supports reseller accounts? Will be better if it haves opensource licence. Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : gsalas@manta.telconet.net www : http://www.manta.telconet.net

Hi, I am investigating an issue with voicemail and realtime. What we are seeing is the following: 1. Caller calls in and goes to an IVR 2. Presses 101 to go to voicemail 3. app_voicemail start and tries to connect to the database trhough res_config_mysql. However, it takes too long to be able to connect (~15 minutes) It seems like it first attemots to connect to the database on 16:25:03 and

Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499014 at X.YYY.32.3:5060;transport=udp SIP/2.0 Record-Route: <sip:X.YYY.32.10;lr=on;ftag=247898> To: <sip:8009499014 at X.YYY.32.10 :5060>;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: "Scott Thompson" <sip:7166359474 at X.YYY.32.10>;tag=247898

I'm monitoring Asterisk with Nagios. Nagios constantly alerts because of too many zombie processes. I eventually had to disable the notification for the alert but why does Asterisk create so many zombie processes, I've see more than 30 at times and it generally stays in the 20s... just seems unusual and wondering if it's harmful, thanks in advance. -------------- next part

We've been asked to quote for a large cluster running Asterisk and our ITSP in a box product. The system will be SIP throughout, with mixed codecs. We're considering using Dell blade servers, 1855 or similar, on the grounds that we normally use Dell machines and they work well, but we need higher rack density. Has anyone used these? Any feedback on whether they're

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

A feature we're often asked for in our ITSP product is to allow customers to upload their own music on hold, or to have it recorded for them by a recording studio with the latest news, weather, etc, punctuated by "Welcome to <customer>, please hold". Since there may be thousands or tens of thousands of customers, and perhaps 10% of customers may want this feature with a

this is a bit off topic, but as there is a growing demand for our sponsor in North America, these can be helpful to others... 2006/3/31, Mike Patterson: > Arnaud Quette wrote on 3/31/06 3:05 AM: > > 2006/3/30, Ian Pilcher: > > >> I will certainly attempt to buy one when the time comes to replace my > >> aging Belkin model, but this is a product category in which mail

Is there some concensus on where to set callerid when making outgoing calls via an ITSP over IAX? Is this best accomplished in IAX.CONF or EXTENSIONS.CONF? Also, tech support at one ITSP told me that the SetCIDName command doesn't do anything. Is this something that might be unique to their server? Or a general statement of fact? Thanks, Michael -- Michael Graves

I have an ITSP we are trying to work with that has an "Unusual" way of working, but that said my understanding of their behaviour is that it is fully RFC compliant. Can someone suggest how I might be able to interoperate under these circumstances: We register fine with them, and send the default asterisk Contact: header of: Contact: <sip:s at x.x.x.x> This then causes ALL

Just been contacted by a UK Enum registrar looking for ITSPs to become resellers of their Enum registration systems ... Is anyone using Enum? Does anyone (other than cynical old me) think that Enum is a spammers best friend? Has anyone received a spam VoIP call yet? (ie. one placed directly over the Internet aimed at a SIP URI to a PBX which allows anonymous incoming calls?) I can see that

Hi peeps, I've been struggling with DUNDi for a few days now and I can't seem to make call from Asterisk A to Asterisk B. If I do a "dundi show peers", it finds the other peer but I can't seem to make any calls. Can anybody help me out here. Here's the situation: Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103 Machine 2: AsteriskNOW --> 192.168.1.69 The

We have been working a OpenSer/Asterisk solution to replace our Avaya PBXs.The OpenSer is to provide scalability and the Asterisk to provide rich features.I know this has been many times for calling card platforms but I'm not sure if anyone has a good scalable solution they are using on their virtual PBX or in a CPE PBX environment?If so I would like to talk to them about buy their deploying,

Hi - I am looking for suggestions for ITSPs for faxing with asterisk 1.8. We are based in the US, so would need an ITSP with US DIDs. #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38

Why not create a listing of the Asterisk resellers. Have a link off the main Digium page and post what asterisk services that particular reseller offers. This way people who are just getting into asterisk know where they can go for commercial support. Maybe the reseller could offer some sort of discounted rate for people who are referred to them through digium? This would boost sales for