similar to: sip.conf with versatel and two NICs very strange problem

Did somebody already configured a Digium card on the network of Versatel in Belgium or the netherlands, and would like to share his configuration. (zaptel.conf / zapata.conf) We have HDLC errors (timings i presume)

Hello, and please forgive the OT question. I'm just becoming desperate. I need two ISDN 30 circuits in Amsterdam, and I can't seem to be able to get a provider. I've tried KPN and Versatel. I'm based in California. Does anyone have any recommendations? Thanks in advance, Jose.

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

hello list, i have a question related to zapata.conf,if i do any change in zapata.conf i must restart asterisk or just i restart zapata ,and how to do . ?service zaptel restart? or there is any other command Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri, dahdi etc. all latest releases.. Sangoma says... driver is compatible with ANY asterisk version... I tried driver 3.5.8... Setup ended with error. I tried (latest) driver 7.0.1 Setup went through, Asterisk is showing dahdi channels... all fine....

When I need to stop a running R script on Windows or Mac I just use the <esc> key which kills the current script and returns the control to R interpreter. But when I run R from JGR the <esc> is useless as well as the other available keyboard keys. Just recently not even clicking on the STOP-symbol (a big red X) placed on JGR top menu bar could terminate a process that had entered a

Hello, I must confess to being an absolute newbie to wine - except for having followed the content of the mailing lists for a while. I would quite like to get involved with the project, and would prefer a testing-oriented or documentation role if there was a need for such. What I'd like to know at this point, if anyone would tell me, is where exactly is the wine project up to in terms of

Hi, because many people contacted me about this the last couple of days and I guess most of them are on this list anyway: - Yes, our new German voiceprompts for Asterisk 1.4 are ready and can be downloaded at http://www.amooma.de/asterisk/service/deutsche- sprachprompts/ - Yes, we are in discussion with Digium about including them into the normal install process. I have no timeline but a

Hello ! Are there any plans at Digium to include also german voice prompts ? Thanks regards Hans

Hi, I have been testing asterisk 1.4 with a view to deploying it in my organisation and I am experiencing jittery voice prompts from the voice mail system. I get this jitter even if I try a simple "hello world" dial plan. I have tried the release of 1.4 and also 1.4 svn and both display this issue. I have also tried it on a dedicated linux box and on a linux install running under

Hi, I am running asterisk 1.6.1.6 with a howler screamer card. I have g729 and alaw trunks from a pbx /sip providers. The howler screamer will only transcode from g729 to alaw / ulaw but my voice prompts are in SLIN and throws errors when i try and access these applications. Is it simply a case of converting the prompts into other codecs and asterisk will pick these up? ? Thanks

I sent this last night but it never showed up in the thread list so I'm trying again. Please pardon me if it duplicates. So I've been banging my head against the rack on this one and am now turning to the group for help. I'm in the process of bringing five Asterisk servers (all originally built from source code by myself) from various versions (1.6.2.x,11.6-cert13, and 13.1-cert2) up

Hi, I am new to Asterisk. I'm using it behind a kamailio sip-router to provide voicemail boxes to sip-users. I followed these instruction: http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+Kamailio+1.5.x to set everything up, using ARA with a MySQL DB. After a few tweaks "everything" is basically working, however, a few questions remain that I could not

Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network

I'm going to encode ~2000 CDs soon. All genres, but 90% of it has distorted guitars... Everything from punkrock to metal to industrial to goth to synthpop to classical to techno to whatever... I've heard that RC2 has some hearable artifacts, even in 192/256 kbps... There have been quite a few "bugreports" since RC2 with people sending samples that even I can differ from the

Hi We interfaced our ASCOTEL PBX to Asterisk. by EuroISDN PRI , DSS1 It works fine on channels 1- 15, but on 17-31 the miststood each other. Asterisk speaks in Timeslots, the PBX in B-channels The signalling is ok, but the bridging is shifted. The first incoming connection is bridged to "nirwana" also no indication is hearable, calling a second internal subcribes bridges them to the