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Does anyone have a copy of spc.exe they could send me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061117/ac0b6a44/attachment.htm

...'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? -- Med Vennlig Hilsen, A. Helge Joakimsen

Dear Andrew, Thanks for your kind responce. Regards, vidura. ============================= > Motherboard with SATA RAID1 support That's a mulit-port SATA controller with RAID in the driver (software). > 256 MB RAM Use a little more RAM. > digium PRI/E1 card Is there any reason you aren't using Sangoma cards? > 1. If I use Software RAID, what would be the impact to my

Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service.

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot of interoperability issues, a common troubleshooting issue was to make sure all endpoints where using the latest version of Asterisk. I have not seen these issues in a while. However I've been working with a customer of mine and this ITSP called IP Communications (IPComms.net) well turns out we have had constant

...son managing the list at > asterisk-users-owner at lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > Today's Topics: > > 1. Re: app_voicemail for spanish (Andrew Joakimsen) > 2. SVN servers down for maintenance (Russell Bryant) > 3. Re: Asterisk 1.4.17 crashing more (Steve Totaro) > 4. Zaptel 1.2.23 and 1.4.8 released (The Asterisk Development Team) > 5. Re: AGISTATUS is SUCCESS even though my PHP script returned > -1 (Matt Riddell) >...

I have some Audiocodes units which appear to be running Linux, according to the unit's own "System Log" kern.warn Linux version 2.4.21openrg-rmk1 #2 Wed Aug 30 17:05:29 IDT 2006 However my contact at Audiocodes claims otherwise On 12/4/06, Yaniv Nizan <Yaniv.Nizan@audiocodes.com> wrote: > > > > I doubt that we are running Linux on the MP-202. Perhaps there is a

Has anyone used the AudioCodes MP-20x? http://audiocodes.com/Objects/Analog_Telephone_Adapter_Series_MP_20X.pdf Seems like a good device, but I can't seem to find anyone actually using them... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061022/6ca85b8c/attachment.htm

...chan_sccp seems to have not been updated in 19 months and how feature-packed is chan_skinny? Has anyone encountered a WiFi phone that is reliable besides SpectraLink? Regards, Andrew On 10/25/06, Martin Joseph <ast@stillnewt.org> wrote: > > On 2006-10-25 15:00:52 -0700, "Andrew Joakimsen" < joakimsen@gmail.com> > said: > > > > > > > Also the Nokia E60 and E61 are hybird GSM/WiFi phones, when you have > WiFi > > coverage your calls will go over that technology and when you aren't its > > > just a regular mobile. Works great if...

...r service anyways, if they would have allowed me time to properly port out the numbers, they would not have created an enemy for life. ---------- Forwarded message ---------- From: Rehan Allah Wala <rehan at supertec.com> Date: Sat, Jan 10, 2009 at 13:56 Subject: Your DIDX account To: Andrew Joakimsen <joakimsen at gmail.com> Cc: "muneeb @ supertec. com" <muneeb at supertec.com>, suzanne at supertec.com Thank You for this email Andrew, Please move your numbers in next 3 days somewhere, we will close your account as per your request on Tuesday. Rehan > On Sat, Jan 3,...

Asterisk 1.2 has no support of t.38 whatsoever, the call will drop before t.38 is ever utilised, not even pass-thru. 1.4 Adds support for T.38 pass through only and no other sort of faxing, the endpoint must support T.38 and you must send your call to a T.38 gateway and you must not use NAT anywhere in your network and you must enable re-invites which could cause CDRs not to reflect the true

...2. Re: Good Commercial Grade Service Provider? (Al Bochter) > 3. Re: sip peer name channel variable? (William Piper) > 4. Re: Good Commercial Grade Service Provider? (William Piper) > 5. Re: Good Commercial Grade Service Provider? (Al Bochter) > 6. Re: BLF on GXP2000 (Andrew Joakimsen) > 7. Re: Linux distro + Asterisk or Trixbox? (Andrew Joakimsen) > 8. Fwd: [asterisk-users] Linux distro + Asterisk or Trixbox? > (Andrew Joakimsen) > 9. Re: Fwd: [asterisk-users] Linux distro + Asterisk or Trixbox? > (Steve Sobol) > 10. Re: Multi Operator (...

When I place a call using Iconnecthere as my sip provider, I hear no ringback when making a call. Does anyone else have this problem or offer any suggestions? Thanks, Kevin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031003/b048f72f/attachment.htm

I have a DID with VoicePulse Connect, but the sound quality is horrible, it is often choppy and the caller's voice cuts out for 2-3 seconds at least once a minute, I have contacted VoicePulse many times, and they do not do anything about it! Does anyone have any similar problems? It isnt my Asterisk config because I have 0 problems using NuFone.

Instead of using a PCI card is it possible to use an outside SIP service for "CO" lines? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030808/33b131e3/attachment.htm

---------- Forwarded message ---------- From: Jeremy McNamara <jj@nufone.net> Date: Jan 22, 2007 5:22 PM Subject: Hater To: joakimsen@gmail.com Continue to Hate on NuFone - Every time you post something people already know that you don't have a clue.

Anyone have an idea how to implement a phone number that can only be called once? The first time it will process normally and any subsequent calls will be rejected.

Does anyone know who maintains the asterisk packages in the openSUSE buildservice? They are not updating Zaptel with their kernel updates and I want to get that matter corrected. I submitted to them a bug report but they seem to not care... https://bugzilla.novell.com/show_bug.cgi?id=407408 ... usually within 24 hours a bugreport is assigned or some sort of comment is made.

On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher > <tilghman at mail.jeffandtilghman.com> wrote: >> It is completely illegal in any country that recognizes patents. > > You mean countries that recognize software patents, right? As re...