search for: suffil

...th 256MB ram. And when my Asterisk is runnig i will setup another at my office and i will connect the two asterisks. Do you think the cpu is big enough? Do you think i can hadle up to 3 simultanus calls on sip with my internet connection? Moritz ----- Original Message ----- From: "William Suffill" <william.suffill@gmail.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, August 07, 2004 8:23 PM Subject: Re: [Asterisk-Users] System Requirements > In sort no. > > Depending how many concurrent calls you do on that system at once you > will hit cpu issues....

I followed the Wiki instructions to get zaptel to work on Fedora core 2. It looked like everything went perfect including the loading of ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy not loading. Take a look at my lsmod...Any ideas? (I am running stable Asterisk on a DL360 - Dual processor) Module Size Used by snd_pcm_oss 46201 0

I''ve just started using CentOS as an alternative OS for some servers for a project. At the time 3.4 was the release of choice. I''m curious how long the CentOS project will release fixes and patched rpms for 3.4 before it would be necessary to migrate these machines to 4.x. I rather know in advance so I can plan accordingly and slowly migrate these over time. I do realize that 4.x

Hi, Reporting from Astricon, Mark uploaded the 1.0 release while giving his speech a few mintues ago.. Bring out the champagne :) Lethol

Hi Folks, I recently setup an asterisk system in order to provide a telephone phone system for my web hosting business at a very low expense. My problem is that DTMF tones are not being recognized when calling the IPKall phone number. Calling my server via FWD and IAXTel works out fine however. Has anybody experienced this with the IPKall service? are they not passing the DTMF tones or am I doing

Could anyone who has successfully configured Asterisk to use g729 to conference 10-20 people please post their configs. I purchased and successfully installed 2 g729 licenses and but when I dial into my conference number on the Asterisk box from a SPA-2000 set to allow all codecs, it always appears to connect using ULAW. My iax.conf file includes the following under the general section

...d greatly appreciated. 1 last thing without starting a flame war who on this list sells the Grandstream BudgetTone's. Yes I know there are probably better options but I need to keep costs down for internal and personal deployments where some other options would be overkill. Sincerely, William Suffill

Hi, I use asterisk for a phone quiz game. I need to store data in a database (MySql, postgres) : telephone number, name (voice), ... and of course the answers at the quetions. What's the best way to store my data ? - script with system() command ? - AGI script - CDR - others ... Thanks Jerome Vous manquez d?espace pour stocker vos mails ? Yahoo! Mail vous offre

How to change available options (behaving) during listening of voice mail? (They are unnecessarily complicated) For example, I don't want to press 3 (advanced options) and again 3 for envelope. I just want to play envelope. Also, when saving message, I do not want to choose folder, I want that message as default be saved in old messages. And, I don't want to press 6 for next message, I do

I found a thread [1] last month about the poor/crappy g729 quality on Sipura units. Anyone noticed an improvement or the quality is still poor? If the Sipura firmware/g729 offers no quality yet, who else is offering a dual channel g729 ATA? I heard about Uniden, but I have no "reports" about their ATA... [1] Sipura g729 call quality to PSTN

...is to G729 (the one that mutualphone is using) won't make it work. I changed the option back again because all other services (FWD, BRI, IAX2) work like this and I don't want to break them. Any suggestions about what I can change to make this work? Cheers! Rene Kluwen Chimit -----William Suffil's Comment----- I've heard problems with the Grandstream G729 and the new digium G729 by MAC ID. Could be a compatibility issue with the implementations. Did you ever use the Grandstream against asterisk with the old Voiceage G729? I've heard that works just fine. -- William This is not...

Thank you to everyone who has offered so far! I've had formal offers from Martin List-Peterson, William Suffil, Greg Varga, Brian Cuthie and Ed Guy (hopefully I haven't forgotten someone....!) Now we just have to decide where the best spot to host it is. What do you guys think? For this week, I don't care if this is a one off. At some point I'd like to have a weekly conference, and if we can...

Hi, I'm working on a RHSA tracking tool, named Sarah. It allows you to build a local RHSA database of different RHEL releases and then allows you to verify systems for compliance (and lists applicable RHSA and required packages). But before releasing my prototype, I would like to know what requirements people have. How they would be using such a tool and what for reports they need to

Ok, After following BJ's advice and removing ogg.so I then got a pbx_realtime.so error in the same fashion. I removed that file, and then the next and then the next as you can see in the log below. I think something is not right. duh here is my sign..lol...but I am not sure even where this ast_register_file_version flag is in a config file or what step I have missed. I am doing a VOIP only

Hey, I am here at Astricon and about to go down to registration. Is there any interest in pictures if I take my digital camera? I am sure that someone is already doing this. (Probably someone official). I would take pictures of each day and upload them to my website if anyone is interested. Let me know! -- Kristian Kielhofner

Hi Guys, Im having a problem compiling zaptel 1.2.4 on CentOS 4.3, anyone encountered this problem before? Here's the error i got: make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o /usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before

Hi Guys, Im having a problem compiling zaptel 1.2.4 on CentOS 4.3, anyone encountered this problem before? Here's the error i got: make -C /lib/modules/2.6.9-34.EL/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o /usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before

...re in the office though. Recently I've been out more than I've been in my office so a VOIP wasn't an option. In this case calls went to me cell and ran up quite alot of minutes. I was wondering how others handle this. Also what carrier you suggest for 2 business cell phones? -- William Suffill

I have a question - if a user calls up w/ blocked caller id I get the following on my phone Incoming call from asterisk This is the same on my Cisco 7940s and Polycom phones. For average users this is not intuitive at all.. I'd like to configure this so if I deploy this at a customer site it says "caller id unavialable". With the spelling done right.... Any ideas on how this

I have backported pbx_dundi.c (rev 1.12 from CVS HEAD) to the 1.0.2 release. The patch has been posted to Mantis at <http://bugs.digium.com/bug_view_page.php?bug_id=0002743>. Jeff Ollie -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url :