similar to: Sip One way audio

I've got a 4 port T1 card in my Asterisk box with a PRI from Qwest as my PSTN interface. I'm experiencing random dropped calls on the various SIP devices I have tested. Network connectivity to the SIP devices looks ok, and I have tried a variety of the devices including all of the following. Grandstream 286 Grandstresm 486 Sipura SPA 1000 Mediatrix 2102 Some example lines from my logs

I've got a client who would love to be able to leave an asterisk server running sompelace, and achieve telephone connectivity using an IP phone from within his Motorhome in his words "I want to be able to work from a mountaintop in Glacier National Park" I've done some initial testing, and a SNOM200 SIP phone comes close enough to working that I have not ruled out the idea as

I'm in the process of turning up a PRI in one of my markets and have run into a problem I have never seen before. I am unable to place a local outgoing call. Long Distance over the same PRI works fine. When I attempt to place a local call using the PRI I see Asterisk attempt to dial, and am greeted with a busy signal. This signal appears to originate on the telco's switch. I have had

All, I was a bit too focused on where I thought the problem was - turns out I wasn't crazy and the dialplan does work as expected. The problem was with dtmf detection - setting relaxdtmf=yes did the trick. Sorry for the premature post for help. Begin forwarded message: > From: Ben Witso <benw@bgwcomp.com> > Date: Mon Jun 21, 2004 7:28:42 PM US/Central > To: Asterisk-Users

I have (more than 1) provider that I receive calls from using IAX, and I have 2 IAX deskphones, all work fine except for some reason with 1 provider, when the call comes in, it doesn't match up with the incomingcall context. (A bit worrying, since I don't want people to be able to relay calls off me.) in iax.conf I have: [ipcomms] type=user nat=yes dtmfmode=rfc2833 host=71.16.179.149

I am just starting out with * so bear with me please. I have tdm400p with 4 fxo modules on it. When I call into the asterisk box from my mobile, I can see the asterisk console picks the call up and routes it to my computer with x-lite. There was no sound coming from either - just silence. I then decided to route it directly to voice mail to see if that would narrow the problem down, but it

On 04.05.2017 10:27, Andreas Oster wrote: > > > Mit freundlichen Gr??en > > Andreas Oster > NOVA Elektroanlagen GmbH > Carl-Zeiss-Str. 3 > > D-76275 Ettlingen > > Tel.: +49 (7243) 5490 22 > FAX: +49 (7243) 5490 54 > aoster at novanetwork.de > http://www.novanetwork.de > > Gesch?ftsf?hrer: J?rg Amann, Claudia Blasi > Registergericht: Mannheim, HRB

try this one exten => 999,1,Answer() exten => 999,2,playback(~.mp3) exten => 999,3,dial (sip/100) exten => 999,4,playbackground(~.mp3) exten => 999,h,Hangup() not sure abt playbackground should be before the dial command or after ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Roland Zagler Sent: Sat 7/2/2005 8:23 PM To:

Mit freundlichen Gr??en Andreas Oster NOVA Elektroanlagen GmbH Carl-Zeiss-Str. 3 D-76275 Ettlingen Tel.: +49 (7243) 5490 22 FAX: +49 (7243) 5490 54 aoster at novanetwork.de http://www.novanetwork.de Gesch?ftsf?hrer: J?rg Amann, Claudia Blasi Registergericht: Mannheim, HRB 361711 Adresse Andreas Oster Am 03.05.2017 um 17:51 schrieb Andreas Oster: > Hi all, > > I am currently facing

I'm having this problem where if the phone is ringing from IncomingCall #1, IC#2 will be immediately sent to VM. Is there somethign wrong with my dial plan? I currently have 4 incoming lines going into a TDM400 with the group set to g0. Could it be that the way I've set this up, if any of the phones are busy, it goes immediately to VM? exten => s,1,Answer() exten => s,2,Wait(1)

I have been wanting something similar. I paid some money for a busy detect routine from newman telecom, but it is not yet done. We'll see what happens. Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Min Hwan Chang Sent: Tuesday, August 09, 2005 6:57 PM To: Asterisk-Users@lists.digium.com Subject:

Am 08.07.2017 um 23:10 schrieb Heiko Schlittermann: > Andreas Oster <aoster at novanetwork.de> (Fr 07 Jul 2017 08:15:05 CEST): >> Hi all, >> >> I am currently struggling with an odd sieve/Pigeonhole issue. Some weeks ago >> I had to replace our dovecot certificate due to expiration. In the past I >> did use a self-signed certificate, but because we now have a

Thanks But this is for demoting a PDC to a BDC. I've done that already. What I want to demote my BDCs to standalone server. I'll give a try to upromote. Kind regards Julien -------- Message d'origine-------- De: andreas oster [mailto:aoster@novanetwork.de] Date: lun. 26/07/2004 16:44 ?: samba@lists.samba.org Cc: Objet: Re: RE : [Samba] Samba as a PDC / Windows NT 4

On Wednesday 09 July 2008 05:40:19 am Andreas Oster wrote: > I have copied the driver files to /var/lib/printers prior to executing > the adddriver command ! The drivers need to be copied to the directory returned by the "getdriverdir" rpc command. On my system the output looks like: rpcclient $> getdriverdir rpc_pipe_bind: Remote machine localhost pipe \spoolss fnum 0x772d

I do not know. But after I did it one time from a windos 10 machine with RSAT my whole dns was broken. The only thing could cure that was to open RSAT on my windows 7 admin PC and delete the entries I added on Windows 10 and save the changes. Daniel Müller -----Ursprüngliche Nachricht----- Von: Andreas Oster [mailto:aoster at novanetwork.de] Gesendet: Freitag, 9. November 2018 11:52 An:

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

So you all know I've been working with Speex on iGlance for ages. Well, I'm happy to report that after all this time, it's done. (Well, as much as anything in this space is ever done, but it works for some large subset of users, and it has a snazzy website to promote it.) For details, please see: http://www.iglance.com To review, iGlance is a P2P VoIP/videoconferencing

So you all know I've been working with Theora on iGlance for ages. Well, I'm happy to report that after all this time, it's done. (Well, as much as anything in this space is ever done, but it works for some large subset of users, and it has a snazzy website to promote it.) For details, please see: http://www.iglance.com To review, iGlance is a P2P VoIP/videoconferencing

Hi, I have just put in a tdm400p with 4 fxo modules and am trying to dial out from x-lite to dial my mobile phone just to test. The output in the asterisk console is like this Executing Goto("SIP/2002-239b", "mobile|61400039953|1") in new stack -- Goto (mobile,61400039953,1) -- Executing Goto("SIP/2002-239b", "localcall|61400039953|1") in new

I've got automon up and recording calls on demand from information I found in the list archives, however instead of ending up with one monolithic file, I've got a -in and -out version of the files in my monitor directory? Anyone have suggestions how I could end up with a monolithic file that does what I want?