similar to: Need feedback on vitelity

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

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

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:

Integrics is pleased to announce version 2.0 of Enswitch, the most integrated platform available for offering commercial telephony services such as ITSP, hosted PBX, calling cards, call shops, number translation services, and much more. Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is effectively the same product as ITSP 1.7. The product has been rebranded as, although it

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

Vitelity seems to be offline to both IP and voice traffic. Is there any place to find out what their status is? Roger Marquis

Any vitelity customers with pbxinaflash boxes? I'm able to call in-house, but failing to make outbound calls. My assigned server at vitelity is not reachable. I can ping to my ISP OK. Any help appreciated. Such as actually how to make email contact with support at vitelity. They're not responding. Thanks, Tom

Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed

Just emailing the list to see if anyone out there has used Vitelity? If so what has been your experience with service, support etc? Thanks Curt

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

I've been considering doing a web based database system, where you can post your termination offerings or wanted, then search by location, price, minimum volumes, etc. I'd probably make it free, supported by advertising my consulting company, or Google Adwords, or something like that. I've got the design written down, all ready to start coding. I could probably have a prototype

I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the

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

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

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

Hi All- ? I have successfully routed calls into our asterisk system from several DID providers in the USA, but for some reason I'm having a problem getting Vitelity to work. ? We are using the IAX protocol, and the symptom is that only about 50% of the calls terminate properly into my asterisk system - the rest get a busy signal.? The ones that do not come in don't show up at all on

I appreciate your report (below), but it's a strange and disturbing coincidence for me. DTMF out through Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? ---- Date: Tue, 22 Jul 2008 12:23:39 -0400 From: "Mark G. Thomas" <Mark at Misty.com> Subject: [asterisk-users]

Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked "sip show peer" and saw that Vitelity for inbound was now reporting "DTMFmode : rfc2833" (it didn't used to), so switched my ountbound dtmfmode to