similar to: Fwd: Asterisk on Solaris 10 (AMD Opteron, Sun Fire X2100)

Hello, world! I'm considering running Asterisk 1.2.4 on Solaris 10 on a Sun Fire X2100 server or two (Opteron CPU, nForce 4 chipset), and apparently this works. I've read that the Zaptel package won't work on anything other than Linux, since it's intended to hook into the Linux kernel in the form of a kernel module. This concerns me, since I've read that ztdummy, the

Hi folks, I am having issues installing Centos 4.3 X86-64 on a Sun Fire X2100(AMD chip). It works with 2GB of ram just not 4 GB. It works with any combo of the 4 1GB chips I have. It only works with 2GB not 4GB any ideas? Thanks, Bill

Hi, We're using gmirror on our sun fire x2100 and FreeBSD 6.1-p10. Some days ago I found this in the logs: Apr 1 02:12:05 x2100 kernel: ad6: WARNING - WRITE_DMA48 UDMA ICRC error (retrying request) LBA=612960533 Apr 1 02:12:05 x2100 kernel: ad6: FAILURE - WRITE_DMA48 status=51<READY,DSC,ERROR> error=10<NID_NOT_FOUND> LBA=612960533 Apr 1 02:12:05 x2100 kernel: GEOM_MIRROR:

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

Hi, Just need to check somthing.. Am I correct in saying that conferencing does not work on a system that does not have a Digium board installed?? Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey

This is not directly * related, but could be. My company is using a VoIP conferencing solution that is suffering from developer neglect. I've considered trying to leverage *, and our internal developers can build the management interfaces. If that plan is not accepted by management, I need to find a commercial solution that works. Requirements: H.323 SIP G7.11/G7.29 Multiple unique

Back in December, there was some discussion here about the Ultra 20''s bios being confused by the EFI labels ZFS uses. After seeing a report about the X2100 exhibiting the same behavior as the Ultra 20, I reproduced the problem and filed: 6407133 X2100 bios intimidated by ZFS-created EFI labels hopefully that bios will get fixed, too. - Bill

I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy for timing. I have now installed the zaptel package and I believe the ztdummy module is loading ok [root@astro asterisk-1.4.0-beta2]# lsmod Module

Hello , I am a newbie to * and have just been able to call a sip User Agent on a different machine thru *. I was trying to set up conferencing between 3 sip useragent on different macines at my worplace but was not able to figure out the procedure. I made the changes in meetme.conf and extension.conf as specified by someone in this mailing list, but * giving some error, " No ISA Tormeta

Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n" "INSTALLED FOR CONFERENCING FUNCTIONALITY.\n" I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What

I was having problems with conferencing when I found a list which suggested that ztdummy might not be loaded. I checked using lsmod and sure enough it was not loaded. When trying to load ztdummy I get an error saying "Can't locate module ztdummy". I am using Asterisk CVS-09/13/03-23:21:19 Any help would be appreciated. Thanks, Chad

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

I do not have any zaptel hardware on the Asterisk box, I could not have meetme functioning. I did modify the Makefile in zaptel directory on line 168 by including ztdummy as one of the modules to compile in. The error message from the concole: -- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

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've been trying to use a set-up whereby I have several TA's connected to an Asterisk server (1.2.4) and they act like they're in a hunt-group i.e. try the first, if busy jump to the next etc. in my extensions.conf I had something like [inbound-trunk] exten => 441234123456,1,Dial(SIP/s1a,20,r) exten => 441234123456,102,Dial(SIP/s2a,20,r) exten =>

Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will

Hi All, kind of need help on the conference module, i'm using freepbx and enabled conferencing, i created a conference number, 6000. when i dial to it, my phone says it is connected but i'm hearing nothing, maybe logs below can help you. also, when i hang up the phone, the conference did not disconnect me. how can i end a conference? thank you -- Executing

I am interested to learn if I need to have ztdummy installed if I do not have any zaptel hardware in my machine? I have found a lot of references with RTP problems which were related to RTP timing (or lack of it). My problem is that voice coming from SIP hardware is OK, but voice going from asterisk to SIP hardware is choppy, full of noise or completely cut-off. Am I going to solve my problem

Does anyone know if the X100P clone cards provide the timer needed to run MOH and the Conferencing service? I have no need for a T1 card, but I'm running asterisk on a dual processor machine with the wrong kind of USB devices, so none of the dummy timers will work for me. Also, I'm running Asterisk@Home (2.4 Kernel). Anyone know if upgrade to the 2.6 kernel will allow me to use the