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After more investigation, I decided to just recompile asterisk (on my newly upgraded Fedora core 4 system). Make dies with this error: "No rule to make target 'usr/lib/gcc/i386-redhat-linux/3.4.3/include/stddef.h" It seems this directory is gone under FC4, and replaced by No rule to make target 'usr/lib/gcc/i386-redhat-linux/4.0.0/include/ I can't find the

Dear, I am looking help with the asterisk pbx, how to setup lynix and asterisk . Thanks -- Talkvoip Telecom Canada Tel:416-893-2089 email: info@talkvoip.ca , talkvoip@gmail.com www.talkvoip.ca <http://www.talkvoip.ca> -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have asterisk up and running now, and installed XLITE on 2 PC's. Both machines (mistakenly) registered as the same user / extension. Strangely, asterisks allows this and both phones can make calls! But, only the first one to register can receive calls at the extensions. 1. Is this normal behavior? (Why allow 2 phones on same extension) 2. Why is asterisk not showing the second phone when

I'm thinking of placing Asterisk on an itx motherboard in a tiny case. The ITX motherboards top out around 400Mhz PII (in terms of power relative to a desktop). How much CPU would I need for an office of 50 people? How much disk storage for voicemail + OS? (typical / average) The system will have no PCI cards (no Digium FSO/FXO cards) - everything over the LAN connection. Thanks,

I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on

...isk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ------------------------------ Message: 2 Date: Tue, 5 Jul 2005 20:39:15 -0400 From: "Michael Stahl" <mstahl@ocg.ca> Subject: RE: [Asterisk-Users] Epia C3 Linux To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <6686D60B6BEA5846B5CEF1BD9C8AF91B1D9E@server.ocg.ca> Content-Type: text/plain; charset="us-ascii" Tak...

I have asterisk up and running now, and installed XLITE on 2 PC's. Both machines (mistakenly) registered as the same user / extension. Strangely, asterisks allows this and both phones can make calls! But, only the first one to register can receive calls at the extensions. 1. Is this normal behavior? (Why allow 2 phones on same extension) 2. Why is asterisk not showing the second phone

I'm using EyeBeam from xten, and whenever I call another user, the callee phone rings but my SIP phone immediately hangs up. The other end keeps on ringing but when the callee answers, there is no sounds. I have found the "Didn't get frame from channel" error occurring in each such call. What does this mean? How can I fix it? -Mike- May 15 22:31:10 DEBUG[4792]:

I'm using the default mp3 files that ship with Asterisk: fpm-calm-river.mp3 fpm-world-mix.mp3 If they were variable bit rate, I think I would see a warning about 'varibel' or similar...is anyone else able to get these files to work? -Mike- ________________________________ From: Sander [mailto:crombeen@rommelweb.nl] Sent: Tuesday, May 17, 2005 6:34 PM To: 'Asterisk

Is is possible to use any of the advanced VOIP features from an analog phone on the analog side of the ATA (like a SPA-2001)? I'd like to use hold, transfer, and conference, etc. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050611/c9cdf4f0/attachment.htm

I have a fresh install of asterisk on a Fedora FC3 machine. (I'm new to unix so forgive the next question if it's obvious). Why does "ps ax" show 6 copes of asterisk running? 3338 ? S 0:00 asterisk -vvvg -c 3370 ? S 0:00 asterisk -vvvg -c 3372 ? S 0:00 asterisk -vvvg -c 3374 ? S 0:00 /usr/sbin/smartd 3384 ? Ss

Has anyone written a LogWatch script for Asterisk? I use logwatch for monitor all my critical services and would like to do the same for Asterisk. LogWatch is very popular, so I'm guessing that someone has created one but hasn't had time to post it somewhere... Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL:

Post your iptables output ("iptables -nvL") and I'll send you the commands you need OCG -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Wednesday, June 29, 2005 9:17 PM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 11, Issue

I haven't heard much feedback yet - anyone here using VocTel? The connection problem turned out to be my firewall, but I'm curious if others experience any voice choppiness or high latency. Some posters have related the problem to specific VOIP providers, some seem to be ISP related (local network latency). Any feedback? OCG

I just upgraded to Fedora Core 4 and Asterisk won't run any more. When launching asterisk, I get "asterisk: error while loading shared libraries: libssl.so.4: cannot open shared object file: No such file or directory". A quick search ("find / -name libssl.so.4") for the file shows the file nowhere on my system. However, when I "yum provides libssl.so.4", yum

I just bough a Sipura SPA-2100 to use with Asterisk. When I use the analog handset plugged into the SPA-2100, the person on the other end can hardly hear me. I check the SPA-2100 setup and their is no mic/spk gain control. Is this a problem with the SPA-2100 or with Asterisk? Any way for asterisk to compensate for the poor audio level (if the problem is the SPA-2100)? Thanks, Mike

My new asterisk install seems to be running fine - including playing all prompts etc without error. However, when placing someone on hold they here choppy music (first second or so) then quiet. I see the errors below. What is causing this? (Note that I am running AsteriskWin32). Thanks, Mike May 17 17:26:02 VERBOSE[3708]: -- Executing Answer("SIP/2434-263c", "") in

I have a SPA-2001 and I didn't realize I could use calling features on an analog handset. Does that mean you can dial *77 and use a VOIP feature? (like forward or hold)? Mike ________________________________ From: Jorge Carrasquillo [mailto:jorge.carrasquillo@gmail.com] Sent: Thursday, June 23, 2005 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

If your phones are setup to connect to the asterisk box by name, then a smart DNS server can just point phones to the backup box after failure. However, since asterisk running on the backup box doesn't know about the phones, this is only half the solution ________________________________ From: Mohamed A. Gombolaty [mailto:mgombolaty@noorgroup.net] Sent: Thursday, June 30, 2005 8:30 AM To:

Any users of the VocTel VOIP service? (Canadian) How have you found the quality (Choppy / smooth audio)? Any problems registering? (I have been unable to register for hours) After reading about the collapse of a big USA VOIP provider, I'm curious Thanks, OCG -------------- next part -------------- An HTML attachment was scrubbed... URL: