similar to: H.323 Cease Fire

Hi, I havn't actually tried this yet, but would it be possible to use asterisk as a mgcp <-> h.323 translator? For example, I have mgcp service from Next Gen telephone company. But i only have a h.323 phone. Would there be a way to the mgcp signalling to hit asterisk, and then have it fire the call out h.323? And vice versa? Just brainstorming. Sean Watkins

Hi all, I am looking for a software package (free or not), or an inexpensive hardware device, which can route calls between an H.323 network and an MGCP-based voice service. Unfortunately, I believe (based on documentation and other forum posts -- I have not looked at the code) that Asterisk can only act as an MGCP Call Agent. Has anyone added code that can talk the 'slave' side of the

Sheesh. I only joined here a few days ago and already there's a flame war. Look, to remove your name from the list is easy. It tells you where to go to manage your subscription down there at the bottom. If you want another mailing list, why not go to yahoo!! or topica and set one up, or set one up yourself. It ain't rocket science with mailman. Even an idiot like me has managed it.

Hi All, At the risk of starting a major flame war and giving Java player-haters more fuel for their ire, I''ve done a performance comparison between Ruby/Odeum and Lucene: http://www.zedshaw.com/projects/ruby_odeum/performance.html Please don''t take this as a "Java sucks Ruby rulez" posting, or that I''ve done any sort of scientific analysis here.

On 05/26/2016 08:45 AM, Valeri Galtsev wrote: > > On Thu, May 26, 2016 5:17 am, Johnny Hughes wrote: >> On 05/26/2016 04:31 AM, Yamaban wrote: >>> On Thu, 26 May 2016 08:00, James Hogarth wrote: >>>> On 26 May 2016 00:57, "SternData" wrote: >>>>> On 05/25/2016 06:43 PM, Always Learning wrote: >>>>>> On Wed, 2016-05-25 at

Hi all, I just installed chan_h323 to interface to a H.323/ISDN gateway. It works really well after two days learning and testing except one thing somebody of you may have an answer to: If I call in from PSTN to a busy asterisk SIP extension I can see a SIP status 486 BUSY, but don't get it passed to the H.323/ISDN side. Asterisk jumps correctly to EXTEN+101 in the dialplan. I tried

On Thu, May 26, 2016 9:30 am, Johnny Hughes wrote: > On 05/26/2016 08:45 AM, Valeri Galtsev wrote: >> >> On Thu, May 26, 2016 5:17 am, Johnny Hughes wrote: >>> On 05/26/2016 04:31 AM, Yamaban wrote: >>>> On Thu, 26 May 2016 08:00, James Hogarth wrote: >>>>> On 26 May 2016 00:57, "SternData" wrote: >>>>>> On 05/25/2016

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Lacy's response in the thread 'Why does everyone seem to dislike *now?', has a small bit that caught my eye. Chan_Skinny made a lot of progress between 1.2 and 1.4, and even more in the later 1.4.X releases. I am curious as to which features/functions that chan_skinny might be lacking compared to chan_sccp. We (the community) now have a small, but active, group of volunteers

Hi, To push voice through a long thin wan (dsl) there are two choices: (1) have the cisco's (7912G) talk g729a to each other (reinvite=yes), or (2) have the cisco's talk to their local * in ulaw (reinvite=no), which talk to each other through a more advanced low-bandwidth codec (ilbc or speex) which is best? (2) would have more latency, wouldn't it? Did I miss a third option?

[Vorbis] Extension proposal - partly serious) In-Reply-To: <Pine.LNX.4.44.0406201942590.21207-100000@sasami.anime.net> References: <Pine.LNX.4.44.0406201942590.21207-100000@sasami.anime.net> Message-ID: <40D70E0E.5090807@ellisfoundation.com> >Anyone else want to take this discussion to another mailing list? We can >hash out a standard there without interference from the

MBvs> On the AD Server i got a problem with the internal DNS from samba MBvs> because systemd-resolver thought it is the DNS boss now. I disabled MBvs> systemd-resolve (systemctl disable systemd-resolved.service) and mange MBvs> the resolve.conf manually. Thanks a bunch for that tip. I was having DNS issues, but hadn't yet gotten to the bottom of it, and this turned out to be

I've got an old HP Netbook, which is just fine for taking with when I'm travelling, to check email and news. I have a very old Ubuntu netbook-remix on it, and it really, *really* needs to be updated to something current. I, of course, would prefer CentOS.... The question is: I see that *if* the specs I just looked up (I'm at work, not home, where I could just turn it on, but they

Hi Philip, Unfortunately, * speaks MGCP only as the Call Agent, rather than as the Media Gateway. MGCP is a master/slave protocol, and it would take some effort to make * work as the slave. I have the same problem: Free Telecom here in Paris includes MGCP service with their DSL. You can call any fixed phone in France at no charge! Rates to mobiles and international are quite aggressive, too.

Hi, I would like to express my appreciation to your big efforts. I am enjoying Asterisk very much. My Asterisk works very well, but I encountered a segfault at voicemail after pressing # to end the recording. Please see the log below. My asterisk is running on RedHat 8. When booting the Asterisk, I found a WARNING around IAX, and it says "Unable to open IAX timing interface: No such

Skipped content of type multipart/alternative-------------- next part -------------- [root@LINUXVM root]# asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= == Parsing

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

Hi, I am running Asterisk CVS from 8/27/04, and since about 8/17/04 Asterisk crashes on reload. I did remove support for h323 (as it crashes my * at random, and I don't need it currently). Here is a cut-out of the last lines when I give a reload command... == Parsing '/etc/asterisk/voicemail.conf': Found -- Reloading module 'app_queue.so' (True Call Queueing) ==

Sam> In France, the second most important ADSL provider (named "Free") Sam> offers a phone line (which uses VoIP but can only be used as a FXS) Sam> with unlimited free calls to landlines. I also have Free ADSL in Paris, and would very much like to get their VoIP working natively with Asterisk. Free assigns each user both a public (for Internet access) and a private (for VoIP

When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found