similar to: Choppiness/Ticking sounds over LAN

OK, background/config. running * (show version reports 0.9.0) on Mandrake 9.2 (kernel: 2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card, no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat reports a minimum of 80+% CPU idle when problem occurs. connect to a Grandstream 101 (GS) via vpn (no nat). Link has 100ms - 150ms ROUND TRIP latency

Hello, We use all SCSI PCI card hardware RAIDs on all 4 of our production Asterisk servers. They all have Digium quad T1 cards and they all have from 2 to 4 T1s hooked up to them. We have had no noticable problems with dropped calls/poor quality. What are you looking to do with this system? what kind of traffic will be going through these 4 T1s? MATT--- -----Original Message----- From: Deon

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Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive Email :

Here is a SIP phone I haven't seen before. Does anyone have any experience with this one? -----Original Message----- From: George Richardson [mailto:georger@netxusa.com] Sent: Wednesday, April 02, 2003 4:56 PM To: clay@ctitec.com Subject: ipDialog Ethernet SIP Phone $199 pad <http://us.st1.yimg.com/store1.yimg.com/Img/trans_1x1.gif>

Hi! I am doing a research about the prices of SIP telephones. If someone can tell me which one are the cheapest and have an acceptable quality... it will be very kind. Best Regards, Mireia

Hello all. I am currently working on setting an S4 domain to replace our aging samba 3 setup. We have found many answers on the net, in various documentation, but when it comes to setting up beyond one node documentation becomes a little thinner. We are setting up a Primary DC with AD, using BIND9_DLZ, also serving dhcp from Primary, and we want to setup a Secondary that is both a

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Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike

Hello, I have a CAC Channel Bank I with FXS cards. I've the system up and running, with just 1 issue. When I make an inbound call, Asterisk says "Zap/26 is ringing", however, the phone never rings. No lights are lit on the CAC during the calll. Outbound call works no problem, and the CAC lights up correctly. Any ideas what could be the problem? -- Richard Cook

Hi All, * is working very well for us now. But I have an issue that I cannot find the answer to - enter guru's!! When our receptionist does a blind call forward I receive the Caller ID, however I do not know if the call is fresh (i.e. ringing in) or forwarded. What I would like to do is to have * prefix the CID External (so that I can tell that it is a fresh call) or Internal (to tell me

Hi All, * is running a dream now, however we have an odd problem that I am sure some guru will be able to sort out for me in no time!! When receiving or making a call about 60 seconds or so into the call we develop choppy/stutter audio problems. It then seems to clear itself only to return again, and so the pattern carries on! This has got me stumped! Our equipment is SipTone II handsets, AVM

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Hey all, I''ve got an interesting one for anyone who''s up for a challenge. Essentially, I have a very choppy effect, that almost looks like timeouts are overloaded or interfering or something, that only occurs when sortables are on the same page as "standard" effects. Here''s what I''m doing: I have a menu that slides in and out on the right side of

Rowland, thank you for the reply. I must have misstated. We have successfully setup our first DC. It works great with DHCP and BIND9_DLZ and updates nicely as it is designed to, kerberos and all. The question is about the second server. Perhaps MY understanding of what I have read on the samba wiki, and others, is different than actual reality. http://bit.ly/2r3IOjt ;) Perhaps if I show

Hi, I've just initiated a new bounty for the above; http://www.voip-info.org/wiki-Asterisk+bounty+callingcard+to+MySQL Any takers or any contributors please respond to me privately. I do not know exactly how the bounty process works, but I can coordinate on this ? SW -------------- next part -------------- An HTML attachment was scrubbed... URL:

Just a little pun there! I've been mostly lurking for a couple of weeks and realize how little I know and understand about this PBX and phone stuff. I did a little looking about and came across a glossary but they terms are -- for me -- kind of out of context. I'm wondering if there is (much as I hate the term "Dummy") a "PBX for Dummy's" or similar. I've

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Hi Hekuran I have installed asterisk-1.0-RC2.tar.gz, asterisk-prepaid-0.3.1 and postgresql. When I tried to call from any IAX client to another IAX client and also sip client to sip client it worked fine. And also the cdr table filled properly. Now I tried to configure asterisk-prepaid-0.3.1 with asterisk. I have compiled asterisk-prepaid-0.3.1 and also copy the configure file. I