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...(compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it should be cheap ;-) Best regards -Thorsten- -- Thorsten G?llner OVM Office Voice Media GmbH Herderstrasse 68 40237 D?sseldorf Tel.: +49(0)211 / 618 57 53 Fax: +49(0)211 / 618 57 54

Hi, I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql database. But with no success. Do you have any hint for me? cat /etc/odbc.ini ------------------ [MySQL-asterisk] Description = MySQL ODBC Driver Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = localhost User = xxx Password = xxx Database = asterisk Option = 3 Port = and /etc/odbcinst.ini

Hello, Limiting the call duration with the L-option of the Dial()-command is working fine, however the announcement is not played. Dialplan : exten => _367,n,Set(LIMIT_PLAYAUDIO_CALLER=yes) exten => _367,n,Dial(SIP/test6,,L(11000,5000,5000)) The call lasts for 11 seconds, but 5 minutes before time runs out an announcement should come. I hear no announcement, not on caller-side nor on

...4. Re: Strange Issue: asterisk deleted (Marie Fischer) > 5. Re: SIP call drops after 32 seconds, but only when.... > (Marie Fischer) > 6. Re: SIP call drops after 32 seconds, but only when.... > (Amit Patkar) > 7. Re: Strange Issue: asterisk deleted (Thorsten G?llner) > 8. Re: Strange Issue: asterisk deleted (Antoine Megalla) > 9. Re: Strange Issue: asterisk deleted (A J Stiles) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Nov 2014 22:08:05 +0200 > From: Antoine Meg...

Hi, I have the following setup: Ubuntu 12.04.02 LTS (64 bit) Asterisk 11.2.1 Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected) WANPIPE Release: 3.5.28 DAHDI Version: 2.6.1 Echo Canceller: HWEC libpri version: 1.4.12 I call via sip into the dialplan. Then I do a "Dial(DAHDI/g1/voicenumber,r)". The call is bridged and everything is fine. "dahdi show

Hi, I am using: Asterisk 11.2.0 libpri 1.4.12 Dahdi: 2.6.1 Sangoma E1-Card with Wanpipe-Drivers 3.5.28 I call my asterisk box via SIP and connect the call to an AGI-Script. Within the script I do EXEC SetCallerPres prohib or EXEC SetCallerPres prohib_not_screened But I get the following error: ast*CLI> == Using SIP RTP CoS mark 5 -- Executing [100 at sip:1]

...omeone is actually and purposely killing asterisk but I do not know what or who is doing that also I know that I am the only user on the system. Again any indicators to solve this very weird issue are welcomed. Regards, Antoine Megalla Sent from my iPhone On Nov 26, 2014, at 6:12 PM, Thorsten G?llner <tg at ovm-group.com> wrote: > > Am 26.11.2014 11:37, schrieb Antoine Megalla: >> Hi, >> >> I am struggling with a very strange issue I have been facing for the past week; >> I have a fresh install of CENTOS 5.11 and I have installed asterisk 1.8.32...

Hi, I am wondering, if there is any tool available, which performs a check for suspicious entries in the dialplan. For example a non existing AGI-Script or a double assigned extension ike that: [context] exten => *100*,1,AGI(test_app.pl) ... exten => 190,1,Answer() ... exten => *100*,1,AGI(never_reached.pl) ... A "normal dialplan reload command" would echo no warning or

After working fine for a week or so my new Quad E1 asterisk 1.8 system has started rejecting outbound calls from the Nortel BMC 450 it is connected to. The cli fills up with these: sig_pri.c: Ring requested on unconfigured channel 255/255 span 3 Is this likely to be a 1) config error 2) cable issue (I made them) 3) hardware problem with the Digium card 4) software (lib pri) Any clues?

We have 2 PRI from AT&T And all is well but only few numbers having following issue. We are getting hangup cause 18 do you guys have any idea ? We have just migrate 1.2 to 1.8 and this issue raised [Jun 7 17:57:10] VERBOSE[23717] sig_pri.c: -- Span 2: Channel 0/3 got hangup request, cause 18 [Jun 7 17:57:10] DEBUG[24856] sig_pri.c: Not yet hungup... Calling hangup once with icause,

...on 2010-09-03 07:47:24 UTC LibPRI: 1.4.11.4 DAHDI: 2.3.0.1 Echo Canceller: MG2 Wanpipe-Driver: 3.5.15 Sangoma-Firmware: 43 (Board A104d with echo hardware chanceller) Typically this problem can occur with echo cancellaltion. But I do not think, that this is my problem. Best regards, -Thorsten G?llner-

Hi! I'm writing to this list because I've got a very confusing issue with our Asterisk 1.8.3.2 installation. On high IO load on the hard drives Asterisk becomes instable and crashes after a few minutes. I tried to reproduce this by running bonnie++ on the hardware while making calls. The calls didn't get disturbed (no noises or crackles) but after about five minutes Asterisk suddenly

Hi there, does anyone have experience with Asterisk-AGI-Scripts in PHP while using pthreads in PHP? Are there any limitations or problems known? Best regards -Thorsten-

Hi, I am using Asterisk 11.5.1. As far as I understood, the following configuration allows a sip client only to receive calls (type=peer) but not to place calls (http://www.voip-info.org/wiki/view/Asterisk+sip+type). Why can I place calls though with this config? sip.conf ... [thorsten] type=peer host=dynamic context=my_context nat=force_rport,comedia secret=... dtmfmode=rfc2833 disallow=all

Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play

Hi, I have Asterisk 11 with DAHDI (Sangoma E1-Card) running on Ubuntu 12.04 LTS. Asterisk and DAHDI-Drivers are installed from source. When doing an "apt-get upgrade" the system packages will be update but sometimes Asterisk is broken. Which packages do I have to exclude when I do not have time to recompile Asterisk/Dahdi each time? libc? Kernel-Packages? Thanks so far! -Thorsten-

Hey Guy, I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me. Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc.. -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, it's me again with a cdr-issue. I have the following example extensions.conf: # dial in exten => 100,1,Playback(hello) exten => 100,n,Dial(local/200,20,rtg) exten => 100,n,Playback(pleasewait) exten => 100,n,wait(10) exten => 100,n,Playback(goodbye) exten => 100,n,Hangup # for local dial exten => 200,1,Playback(hello) exten => 200,n,wait(10) exten =>

...isk but I do not know what or who is doing that also I know that > I am the only user on the system. > > Again any indicators to solve this very weird issue are welcomed. > > Regards, > Antoine Megalla > > Sent from my iPhone > > On Nov 26, 2014, at 6:12 PM, Thorsten G?llner <tg at ovm-group.com > <mailto:tg at ovm-group.com>> wrote: > >> >> Am 26.11.2014 11:37, schrieb Antoine Megalla: >>> Hi, >>> >>> I am struggling with a very strange issue I have been facing for >>> the past week; >>> I hav...

...ws 2 locks. The blue lock shows "Signaling is encrypted using TLS" and the orange lock shows "Media is encrypted using sRTP". BUT i hear no audio. After ~60 seconds I get the following message: NOTICE[21005]: chan_sip.c:28800 check_rtp_timeout: Disconnecting call 'SIP/tgoellner-0000002c' for lack of RTP activity in 62 seconds "sip show peers" shows me, that my Blink-Client is registered on port 60071. All other SIP-Clients (no TLS an no media encryption) are registered at port 5060. I tried to open the tcp and udp port range from 10000 to 61000 (in ipta...