similar to: MoH trouble with latest bristuff (0.3.0-PRE-1f)

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

Hello, Using asterisk-1.2.1 I am trying to convert my music-on-hold files from .wav to alaw: % sox moh.wav -r 8000 -c 1 moh.al resample -ql The file sounds fine when listened with: % sox mox.al -t ossdsp /dev/dsp But when listened through asterisk with an alaw SIP phone the sound is clicky and too fast. Did I forget something in my conversion command? -- ldm@apartia.fr

I looks like someone decided to bundle a patch that hasn't been merged yet. Good for testing, not so good for initial impressions. In /etc/asterisk/asterisk.conf add or uncomment this: [options] ;silence_suppression=yes And see if that helps. You need a timing source for it to work, which is why it is disabled by default, but the logging might be a bit chatty in any case. Dan

Hi, I am having a difficulty with getting two realtime user?s to bridge on answer. I have managed successfully to bridge the same two users/channels via the Bridge Manager api command and confirm that the two communicate directly bypassing the asterisk server (I confirmed this with Wireshark). Does anyone have some ideas? I have put some log entries below. I haven?t attached my

Bristuff has been updated; http://www.junghanns.net/downloads/bristuff-0.3.0-PRE-1u.tar.gz -- Vidar

Hi, I am trying to upgrade an Asterisk 1.0 with chan_capi 0.3.4 to a more recent version, but I cannot find any working combination of Asterisk an chan_capi any more: On "junghanns.net" there is a chan_capi 0.3.6, but this won't compile against any recent Asterisk (missing channel_pvt.h). The production version of BRIstuff comes with an old asterisk (1.0), the experimental version

Again trouble compiling bristuff-0.3.0-PRE-1q with the florz patch on a x86_64 box (I guess nobody is using x86_64 platform or is able to fix this themselves?) First of all when bristuff is downloaded and compile is started it appears that the bristuff Makefiles are badly broken. The asterisk Makefiles all do see to find the kernel sources on a RHEL4 box in the proper directory, the pure

Hi, can anybody tell me what the errors mean and why my asterisk server falls from time to time. From time to time means several hours, not regularly. I also can provide a core if someone can debug? Thanks and regards Jan 11 14:34:59 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83, tei 64 Jan 11 14:34:59 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:03

Hello, I am trying to install a Billion ISDN on Asterisk I have Debian Etch and I installed theese packages: apt-get install linux-headers-`uname -r` apt-get install make apt-get install ncurses-base ncurses-bin ncurses-term apt-get install libncurses5 libncurses5-dev apt-get install bison apt-get install openssl apt-get install libssl0.9.8 apt-get install libssl-dev apt-get install

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

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 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

Hi All, I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Thank you in advance, Dominik

Hi, Reading from www.voip-info.org, i can see that "Junghann's chan_capi is now part of bristuff <http://www.voip-info.org/wiki/view/Bristuff> , as of version 0.3.0-pre". What does that really mean ? Shall I understand I can share a Junghanns QuadBRI board between 2 CAPI-enabled software (like a 0.3.0-pre bristuffed Asterisk for instance) ? If positive, what are the pros and

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

Hello, I am trying to install bristuff-0.3.0-PRE-1x.tar.gz on debian with kernel 2.6.19.2 and I've got some errors connected with XPP. I was wondering if somebody managed to install bristuff with this kernel or any kind of kernel 2.6.19. The bristuff mentioned above contains zaptel 1.2.10 not 1.2.12. -------------- next part -------------- An HTML attachment was scrubbed... URL:

has anyone seen a bristuff version compatible to the actual *1.2.6/zaptel 1.2.5 ? the actual bristuff-patches version 0.3.0 PRE 1l doesn't apply correctly anymore...

Hi I've got a few 8 port Junghanns BRI ISDN cards. Dialling in and out is working fine but the Telco's busy or invalid number indications are not being passed through to the user. I have priindication=passthrough in my zapata.conf but this doesn't seem to help. I'm using Asterisk 1.2.13, Zaptel 1.2.10 and Bristuff 0.3.0-PRE-1v. This is happening on three different boxes that

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