similar to: Scratchy audio on Bridged PRI Calls

P.S. I _had_ read the other posts that suggest changing to unicast sntp mode. This didn't help. I eventually setup a new ntp server on my LAN, and used it as a sync source. Everything seems OK now. Obviously a problem with my other ntp server. Cheers. -------- Original Message -------- Subject: Cisco 7940 - Disappearing Clock Date: Thu, 28 Jul 2005 11:50:35 +1000 From: Rod Bacon

I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be "partially screwed". -- ==========================================

I posted something on this a week ago, at which time I was told that this was an 'old' issue. Since then, I've spent hours looking, but can't find the answer. For some reason, some of my CDRs (both to CSV and MySQL) are being written with the wrong callerid. As best as I can determine, they are being written with the CLID of the _last_ caller to access the specific ZAP channel

I've posted this to SNOM, but was wondering wheter anyone here has issues with SNOM 190 phones not showing the correct DST adjusted time (using the latest firmware). -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 ? ?Fax: +613 99401650 FWD: 512237 ? ? ? ? ? ? ? ? ?

If you see a wj0011 version of firmware for Zyxel Prestige 2000W floating around (I found it in a German forum), KEEP AWAY. It completely trashed the wireless networking in my phone. -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237

Will the CVS HEAD version of the Zaptel drivers work with the STABLE branch of *? -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 FWD: 512237 ICQ: 5662270 ==========================================

I have several * servers, and have noticed something most annoying on recent builds. On an old server, running a CVS 1-0 checkout from 10th May, incoming zaptel calls (ISDN PRI) start processing the dialplan immediately, and look something like this... "Accepting call from 'XXXXXXXXXX' to 'XXXXXXXXXXXX' on channel 0/2, span 1" My other servers are running either the

Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1)

Can anyone tell me if Asterisk would speficically benefit from the reduced latency of a preemptible Linux Kernel? I know it was recommended against in the early days, but I'm wondering if there are any updated opinions? -- ========================================== Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600

I anyone has any hesitations in upgrading their 405P (or 410P) to V2 of the firmware, read below; I installed one today (turnaround time around 2 weeks to Australia, inc. economy freight in both directions... impressive!) and have noticed immediate, significant improvements. Audio levels are better (have set tx and rx gains back to 0.0) and missed frames have gone (popping, clicking,

I've been using Asterisk for over a year now, and think I've pretty much got it nailed form a voice perspective. We have just purchased a couple of Video phones to start experimenting with IP video, with a view to eventually building an IP media platform, such as Intel's HMP. I have record/playback working in Voicemail, but am wondering what status Asterisk's AGI (specifically

I apologise in advance if this is a silly question, as "legacy" telephone technologies are really not my forte. Is there an E1 card that can provide clock source? (E.g. Make my asterisk server look like a telco to my legacy PBX system?). What I am trying to achieve is the following: --ISDN-------| Asterisk |-------ISDN--------| Legacy PBX |-- --

I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to

Hi all We run an * 1.2.4 under FreeBSD with ztdummy kernel module. zttest reports 99.9something % of accuracy, so timing should be fine. SIP connections work fine, but we have a strange problem with IAX2 connections. When an IAX2 call originates from the FreeBSD Asterisk to another Asterisk, the sound is scratchy (sounds a bit like a 50Hz ground loop). It's not a problem of the

The one I like: http://www.rhetorical.com/cgi-bin/demo.cgi is toast. I think they went broke or got aquired by someone. Also, is there a Festival voice that sounds as good as Rhetorical or the AT & T stuff? The default one is barely legible. Since Festival is a little brutal to configure, I'd like to get someone's recommendation then go through the pain of reconfiguring it only once.

Nobody has been able to answer this. Not even Digium at this stage, but I'm hoping someone here, smarter than I, will be able to. We are running some TE406P (upgraded 405Ps) cards performing mainly PRI bridged calls. After a server is brought up, calls sound absolutely perfect. Over time, delay (latency) creeps into the calls. What is really weird about that is apparently with the new

asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were "scratchy" and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm trying to call them now, but it's Sunday morning in the sticks, and my chances of

Hi, Has anyone tried to use gnokii to send/receive SMS messages via serial or USB with AT commands while running Asterisk? Some of my calls have a "scratchy sound" once in a while. It doesn't seem to be due to packet loss but some kind of interference (CPU is ok, etc.). I've noticed some coincidence in time between this scratchy sound and the gnokii process. I have a bash

Dear All, I have move from the USA to Sydney, Australia. I have gone from a data center environment at work and cable at home to a 513k/128k ADSL line. I'm experiencing two issues; 1) There is a latency of .5 - .8 seconds between me and the USA. 2) I have been in two calls where my voice has been describes as 'Scratchy'? I'm using a SIP Phone from SJ Phone, and a Plantronics

I'm currently working out the config bugs on my * box and I'm noticing that the meetme is very scratchy. As in not usable scratchy tho I can hear the audio it sounds like when you talk through a fan. Anyone have any ideas? Linux 2.6 with RTC installed. Using stable release and SIP devices. -Don