similar to: Help me ($$$) with install h323

Hello * users I've problems with sound quality on zaphfc Asterisk works fine good sound quality. If I do "make load" in the bristuf.xx zaphfc dir then sound quality drops directly. Even if I don't load the chan_zap in the modules.conf I use this config on more (even old 400Mhz machines) and works correctly. Looks like an hardware problem but I can't find it. I don't

Hello * Users Did somebody get managed to get AreskiCC work under mysql. If so is there anywhere to find the database structure for mysql. Thanks Sjaak

Hello I like to make calls to an h.323 device. I'm using Nuphone h323. Compiled everything okay "I Guess" When I make a connection * SIP > h323 device, the phone is ringing and then * tells me "No one available....." and disconnect Thinking this is a codec problem and check in CLI h.323 show codecs and * shows nothing. I try many combination in the h323.conf like.

Hello I like to call to 2 providers provider X = IAX provider Y = SIP exten => _06.,1,Dial(IAX2/X/${EXTEN},30,r)&(SIP/${EXTEN}@Y.Y.Y.Y) exten => _06.,2,Hangup Provider X is working but provider Y never shows up. What's wrong ?? How can I get provider Y working a fraction earlier the provider X Thanks Sjaak

Hello everyone I'm using rsync over ssh rsync -ave ssh bla@bla.tld:/home /backup/server1 This works great for many years now Now I'm playing with Nagios and NSCA but how can I detect if rsync has done everything well. nsca works simple <hostname>[tab]<svc_description>[tab]<return_code>[tab]<plugin_output>[newline]. I have a text file named backup_okay with

Hello This Question maybe ask many times. I'm From the Netherlands and looking for an sip/aix > pstn provider in the US.I asked wipphone, vonage but they don't work for europe. I've visit many not finished website's about sip > pstn Can somebody recomed me any good sip/aix provider in the us. Thanks

Hi all Xapian-devel, Gist: https://gist.github.com/10d2222d8bffe8d7631d I'm using Xapian-TermGenerator to extract Norwegian sentences to vsm (vector space model) using TermGenerator. But when I test generating vsm from 'Truet med ? stevne misforn?yd PC-kunde - PC-leverand?ren Asus likte sv?rt d?rlig kundens misforn?yde leserbrev.' It doen't return 'asus' result in vsm.

Hi, I'm trying to setup a small BRI ISDN <-> voip gateway. The ISDN card is based on Cologne chipset, so I try set it up with zaphfc. The versions i'm running: kernel-2.4.27 Asterisk 1.0.7-BRIstuffed-0.2.0-RC8e zaptel modules 1.0.7 zaphfc is from bristuff-0.2.0-RC8e When I'm doing the insmod on zaptel, zaphfc, zaprtc: Zapata Telephony Interface Registered on major 196 PCI:

Thanks in advance for your patience and assistance. Since a system administrator's work is most successful when nobody knows you've done anything....(ugly that...you get neither raises nor equipment funding as a result), I prefer to learn from other's experiences rather than fail in the same pits. We are considering a switch from UW to Dovecot, since our users persist in using

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make

I don''t think that modelling domUloader (or pygrub, for that matter) is going to be optional for VSM 1. Do the CIM providers have any plans for this? Thanks. Andrew _______________________________________________ Xen-users mailing list Xen-users@lists.xensource.com http://lists.xensource.com/xen-users

Hi! Asterisk in Debian/Lenny claims to be bristuffed, not? At least the the Debian patch tracking system shows the bristuff-patches: [1] http://bit.ly/bRRHe7 We have a QuadBRI-Card and recently needed support from Junghanns.net but they refused telling us there is no bristuff installed because of the show version output: *CLI> show version Asterisk 1.4.21.2~dfsg-3+lenny1 built by pbuilder @

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup

I have a problem with CDR recording in Asterisk 1.2.x. This is the situation: An Asterisk 1.2.0 (BRIstuffed-0.3.0-PRE-1 with florz) machine with a single HFC-S ISDN BRI card. I log the call records to both the Master.csv and MySQL. The problem is that when an incoming call from the ISDN line is logged to the CDR, the "src" and the "clid" field show up as something like

We're having a horrid problem with our asterisk setup. Sometimes calls just go dead - we can't hear what the other end is saying. (I think they can't hear us either). The call doesn't hang up until one of the callers gets bored. Internaly we use Thomson ST2030 SIP phones. Externaly we have 3 ISDN BRI lines (6 channels total), connected to an Eicon Diver server card (4BRI).

Hi, I'm using spandsp-0.0.3 [http://www.soft-switch.org/downloads/snapshots/spandsp/ spandsp-20061116.tar.gz] on a bristuffed asterisk (1.2.13) [http://212.91.251.199/~junghanns.net/downloads/bristuff-0.3.0- PRE-1v.tar.gz] libtiff is at version 3.6.0 Running on: Linux router2 2.6.17-2-686 #1 SMP Wed Sep 13 16:34:10 UTC 2006 i686 GNU/Linux Debian testing distro. I've tried many

I have a problem with Asterisk-bristuffed using a zaphfc card. I am located in the Netherlands, so I have an ISDN line from KPN. When I start Asterisk, and plug in the ISDN line, everything works perfectly for about 5 minutes. And then the ISDN line is down for 1 minute, and after that minute, the line comes back up and works for another 5 minutes. Every time the line goes down I get the error

Using the latest(?) bristuff (Asterisk 1.0.4-BRIstuffed-0.2.0-RC3a) I have problems with loosing the D-channel. Most of the time, after the message "PRI D-channel down" it only takes a second or so to come back up, noted by the message "PRI D-channel up" However, today most of the time the D-channel stays down. Calls come in, but can't be answered. Does anyone know

I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been

Hi all, My BRIstuffed 0.2.0-RC7k is running fine on my debian box for voice calls over ISDN2. Now I want to implement receiving incoming faxes into my setup so I did a google and some reading on the wiki. I got the spandsp 0.0.2pre10 package compiled and installed, patched asterisk's apps makefile and compiled * again. This all worked out fine. When integrating the RxFax into my dialplan the