similar to: Asterisk Crackly Bad quality

-------------- next part -------------- An embedded message was scrubbed... From: Deon Rodden <drodden@webunited.net> Subject: Re: [Asterisk-Users] Strange Quality problems with Asterisk, Gentoo, Redhat and Kernels - /dev/dsp Date: Thu, 30 Sep 2004 09:05:39 -0400 Size: 5509 Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20040930/289c69cc/dsp.eml

I've compiled Asterisk on Redhat 9 and Fedora Core 1 in the past, generally without any problems. Especially w/ the stock kernel, which I generally loathe. When I tried to upgrade my Redhat 9 Server to the 2.4.27 kernel, doing a manual/clean compile, I had massive quality issues. I was forced to go back down to a stock 2.4.24 kernel. Never figured out why. Now, I've installed Gentoo

Hi to all, I already setup asterisk on REDhat 9.0 linux machine. I will have 4 physical phone lines and 10 IP phones for it to use. I have a network setup already. Is getting TDM400P - 4port FXO from digium enough to start? Do I need anything else? Thank you

I've got two Polycom 500's that I'm playing with, and I want view the status of either phone, (busy/on the phone/etc.) from the other. I've got this cute little 'Buddies' button, and I can add contacts to that. But the status doesnt actually update. Do I need to setup realtime for asterisk? Can anyone point me to some documentation or give me some hints? Thanks, Nihal

Hello, My name is Nihal. I am a 4th year, CSE undergraduate student at National Institute of Technology Surathkal (NITK). I am proficient in C/C++/Java/PHP languages. Looking through GSOC organizations and projects. I came across Xapian. I have looked through the project list for GSOC'17. Improving Xapian's PHP Bindings project looked right for me. I have been reading up on the resources

I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>

Hi, I don't have a zaptel device for conferencing. I read from the lists, that ztdummy and zaprtc need to be installed to get conferencing. I could able to compile successfully with ztdummy and when i receive the call it says, -- Goto (13732,s,1) -- Executing MeetMe("SIP/-08118800", "1234") in new stack == Parsing

Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=0000875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machine that's never had any * on it before, and has a 2.6.5 kernel with a matching

I finally got 2.4 recompiled with RTC as a module: Module Size Used by Not tainted autofs 13684 0 (autoclean) (unused) acenic 241092 0 (unused) iptable_filter 2412 0 (autoclean) (unused) ip_tables 15864 1 [iptable_filter] e100 62340 1 rtc 9084 0 (autoclean) Here is my

Does anyone know where to disable rtc support on redhat 9.0 using make menuconfig? I thought I disabled it but still got the following error when trying to make zaprtc: zaprtc.c:109: storage size of `rtc_irq_timer' isn't known zaprtc.c:719: storage size of `rtc_fops' isn't known zaprtc.c:107: warning: `DECLARE_WAIT_QUEUE_HEAD' declared `static' but never defined make: ***

I have a system with no Digium hardware in it (two others with 2 X100P cards in each of them as well). I'm interested in using MeetMe in the one without the hardware (it works great in the ones with the hardware). I can't use ztdummy, because the system has usb-ohci drivers, rather than usb-uhci. I have read the little there is about zaprtc, and I am wondering whether there is a

zaprtc 'load' is failing on my machine (the make was fine, same output as other posts to this list) [xxx@xxx zaptelrtc]# make load sync modprobe zaptel insmod ./zaprtc.o ./zaprtc.o: init_module: Input/output error Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or

Any help would be appreciated as I am a novice trying to work around a difficult situation. This is what the zaprtc helpfile says: zaprtc, getting zaptel timing out of your realtime clock ======================================================== Make sure that you _dont_ have rtc support compiled into your kernel! INSTALL: make USE: make load REMOVE: make unload I interpreted this as

This is a bit off topic, but I'm not sure where else to start. I've got a friend in Australia who works for a big ISP there. I guess they want to provide VOIP service to their customers. Can anyone give us an idea how someone can, I guess, purchase phone lines to sell as VOIP lines? I'm a bit lost here, but basically what is needed to become a VOIP provider? Would you buy a bunch

Hi I want to set up * as a conference bridge. I would like to be able to conference is SIP calls (up to 12) I am looking through all available documentation for * to get info on how it is done. No luck so far. Can somebody direct me to the info in this subject? Thank you Serge _________________________________________________________________ Protect your PC - get McAfee.com VirusScan Online

Hi, I'm trying to intall zaprtc on my machine (I don't have UHCI_USB nor Zaptel) and I have a strange behaviour during loading. When I do >make load I have : ./zaprtc.o: init_module: Input/ouput error With a quick dmesg got: rtc: I/O port 112 in not free Is someone have any idea to solve this ? I saw on the README file that rtc support should not be enabled in the kernel, but I

I feel there is some unclarity about the need for a timer in Asterisk. As Asterisk moves into production environments, this needs to be sorted out. I'm setting up an Asterisk in a production environment with no Digium cards at all. It is planned to be used as a clean SIP hosted PABX, where we connect to the PSTN via gateways with proper echo cancellation. There is no USB driver on the card.

I have compiled the zaptel library and zaprtc on a system that gives the following from "uname -a": Linux fxxxxxx76.mydomain.de 2.4.18-64GB-SMP #1 SMP Wed Mar 27 13:58:12 UTC 2002 i686 unknown Makefile for zaptel had the following line uncommented: # KFLAGS+=-D__SMP__ When doing the "make load" for zaprtc I get the following error: modprobe zaptel

Hi, i am testing stuff for a couple of days now with Asterisk CVS-v1-0-07 and everything worked fine sofar when suddenly the voicemail and musiconhold sound output stopped working. The voicemailmenu still works though. I can see the voiceprompts etc in the debug messages on the asterisk CLI but i cant hear anything. Everything else works fine though. I can call out fine etc. I did some network

I need a timing device for the DL360G2 for conferencing and meetme. For a timing device I have 2 X100P cards but neither will work in my DL360G2. The system will not even boot with either card in the system. Other PCI cards seems to work fine. I called Digium support and was told that there must be a conflict between the card and my Compaq DL360G2. I then moved on to ztdummy. I'm sure the