similar to: Vitelity and AsteriskNOW

I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal

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

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

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]

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

FYI/Heads up, I /just/ received what looks like a phishing attempt for information about Open Source PBX usage. It says it comes from Digium but all the links (including the one for digium.com) point elsewhere. Rod --

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

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

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

Hi, I currently have some systems on AsteriskNOW 1.7 & have been happy with its clean simplicity & reliability. Are many people here using AsteriskNOW 2.0.x? How do you feel about it? Did Digium stick with their previous philosophy of keeping everything very vanilla & making it clean & simple for someone who understands how to manage CentOS, FreePBX, tftp, ntpd, etc. but

I have a rather simple setup running under Asterisk 1.4. I'd like to move it to a new install of 1.6. Before I bring it online are there any gotchas I should look for? A Gotcha README would be better but searching with Google and the forums, for me, gets hits that deal with hardware issues -- cards etc. Nothing about depreciated/changed commands. TIA, Rod --

When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays a message at the top "Your browser is not supported by this version of GUI!", and "We recommend using Firefox". Does this mean that it is known NOT to work under IE7, or just that it is insufficiently tested to be guaranteed? It's easy to ask techies to use Firefox, but if we are trying to sell a

I'm still searching but hoping someone can offer a clue-stick. Long story short! I had a server crash suddenly and all I can get at are the files. Built a new host and copied the data and config files over, correcting ownership and permissions (hopefully) as I went. But now I can't get logged in. Messages in /var/log/dovecot/dovecot-info.log, without saslauthd running, are like

Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070324/8bb0be73/attachment.htm

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

Guys, I've downloaded AsteriskNOW few days ago so I'm new to this product. The first issue is on service provider area. I've already used a VoIP account already configured with my ISP, it works fine! This configuration has been used until now with the client SJphone, Now I would use this profile as main VoIP service provider to setup in AsteriskNOW. Here are the profile detail as

Sending this to asterisk-users, in case anyone has AsteriskNOW experience they can share. Joe ---------- Forwarded message ---------- From: Joe Wood <schmoe at gmail.com> Date: Wed, Aug 18, 2010 at 9:22 AM Subject: AsteriskNow REGISTER'ing s@ extension for all inbound trunks To: asterisknow at lists.digium.com Hello. Can someone tell me why AsteriskNow is reverting to registering

Guys, please help me in understanding what I'm mistaking... Description: I've configured my AsteriskNOW (beta 6) server, in service providers section, with the parameters provided by my ITSP. Until now I've used this configuration with SJphone and all worked perfectly. Now I've decided to use this account with AsteriskNOW to begin my experience with a VoIP based PBX. The

Hello, Is it possible to set things such as parts of config files are edited though AsteriskNOW GUI while other parts remain "hand editable" ? AsteriskNOW website include screenshots but not much information (such as user manual) beside that. This thing was the one that kept us from using freepbx (let me say I don't mean it's not possible with freepbx : I mean we couldn't