search for: kwem

Karsten Wemheuer <kwem at gmx.de> schrieb: Hi Karsten! > the timeout value of 15 minutes directs me to an issue with session > timer. Try to refuse them by putting the line > session-timers = refuse > into the general context of sip.conf. Reload the sip stack with "sip > reload". So...

Hi list, in asterisk 1.4 and maybe earlier it was possible to use voicemail system with mailboxes starting with some special characters like *. The line in voicemail.conf was like this: *123 => , AB,,,tz=cet|attach=no| Calling exten => s,n,Voicemail(*123,su) is working in asterisk 1.4. In Asterisk 1.8 the above scenario is not working any more. The Voicemail application reports an

Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with "core show applications" Do I need to install mISDN or other modules for using that ? Regards M.Shirazi

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

Hi list! My Problem: all calls to international numbers will be dropped after exactly 15 minutes... I have a VoIP-account by Deutsche Telekom. This is what I see when I call someone (my parents) and the connection will be dropped: == Using SIP RTP CoS mark 5 -- Executing [+39015222222 at default:1] Set("SIP/00493511111111-00000125", "newNumber=0039015222222") in new

Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don?t in the initialization... I have in my file that comands: touch /var/lock/subsys/local modprobe zaptel modprobe wcfxo safe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody

...rs@lists.digium.com> > Subject: Re: [Asterisk-Users] rc.local dont works > Date: Mon, 26 Jan 2004 07:33:24 -0200 > Reply-To: asterisk-users@lists.digium.com > > Ok! > > Thanks > > miklos > > ----- Original Message ----- > From: "Karsten Wemheuer" <kwem@gmx.de> > To: <asterisk-users@lists.digium.com> > Sent: Saturday, January 24, 2004 9:42 AM > Subject: Re: [Asterisk-Users] rc.local dont works > > > > Hi Miklos, > > > > listas iPfone wrote: > > > Hi ! thanks for the answer.. > > > > &g...

...Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------ Message: 4 Date: Thu, 04 Oct 2007 14:29:01 +0200 From: Karsten Wemheuer <kwem at gmx.de> Subject: Re: [asterisk-users] Configuration files inside SQLite3 To: asterisk-users at lists.digium.com Message-ID: <1191500941.1267.4.camel at baghira.temp.temp> Content-Type: text/plain Hi Mark, Am Mittwoch, den 03.10.2007, 11:15 -0500 schrieb Mark Michelson: > GNUbie wro...

Hi Neil, Am Mittwoch, den 17.12.2014, 09:08 -0500 schrieb Neil Cherry: > Doe anybody know of a way to redirect both channels from a bridge to > different dial plan extensions from the using the AMI. > > Currently, as soon as I redirect one of the channels the other appears > to be dropped and gets reorder tone (congestion, fast busy). > > I guess what I really need is a

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem

Hi Luca, Am Samstag, den 13.06.2020, 08:28 +0200 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider > is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to understand what my > partner sayd... > > It can NOT be a problem of other

Hi, I've been searching through the wiki but did not find the solution, so I hope this is not a faq... The class of MoH is derived from the definition in the channel of the originating call, as far as I understand. But is it possible to define classes other than the default for a) MoH during parking calls and b) MeetMe Conferences (the music played if You are alone in the conf) Thanks

Hi, I just wonder, if it is possible, to suppress my own number on outbound dials with chan_capi. I took a look into the sources and think it might work with toggeling the "@" in front of the outbound msn in the dialstring. (Dial(CAPI/@msn... vs. Dial(CAPI/msn... But it doesn't work. Maybee I'm wrong and misunderstood the code. Thanks for any answers! Karsten

Hello, I would like to install the current beta1, but I can't find a working bristuff package. On 'junghanns.net' I found two versions, one for use with the stable (1.0.x) tree and one for the CVS. As far as I understood, the CVS version should be the correct one. But when I try to use the included libpri-patch to patch against the libpri 1.2.0beta1, it doesn't work. Any hints?

Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature "partial rerouting" which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user

Hi, I'm relativle new to *, so I may be wrong. I build up * from cvs today (show version: CVS-01/15/04-16:27:36). In an test I use 2 SIP phones (linphone) to connect to eachother. The phones are called via the extensions 100 (user 'kwe') and 200 (user 'phone'). I can call from one to another and I can park a call and take it again: 1) call from 100 to 200 2) press # and

Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but

Hello, while testing the following scenario, I ran into trouble: One * box with two AVM active controllers in Point-to-Point-Mode is connected to another * box with ZapHFC/Quad-BRI cards using bristuff in NT-mode. All is working fine, I can call from one box to the other and vice versa. But if I'll cut one line, it is not possible to place an outbound call from chan-capi accross the still