similar to: Delaying SIP disconnect after incoming call hangs up?

Hello, just a short hardware question. Does CentOS 5.3 supports a Leadtek LR2960 (model S26361-D1910-V128, agp, 128mb) graphic card with a nVidia GeForce FX5200 chip and dual dvi? Thank you very much! regards Olaf

Here's a patch against 4.8-RELEASE kernel that allows disk writes on softupdates-enabled filesystems to be delayed for (theoretically) arbitrarily long periods of time. The motivation for such updating policy is surprisingly not purely suicidal - it can allow disks on laptops to spin down immediately after I/O operations and stay idle for longer periods of time, thus saving considerable amount

Does anyone know if someone makes a hard video phone for SIP. Dave

I have a TDM400P with 2 FXO cards and I'm using Asterisk@Home 2.8 I noticed that when I place a call to the analog lines from outside, Asterisk takes a while to actually ring the extension the call is being sen to. I've been doing some tests, calling from my cellphone and here is what I see... - After the first ring on my cell, Asterisk logs to the CLI that is has an incoming call -

Hi, A friend has a few hundred deployed LeadTek BVA8055's and needs to bulk re-provision them. There isn't much documentation on the web. Anyone have documentation explaining the LeadTek provisioning process and the provisioning file format? -- Eric Chamberlain

As I've said before, I have a number of phones that I've converted from Packet8's Virtual Office to work with my Asterisk stetup. They are basically Leadtek 85XX SIP devices with Astra 390 phones connected. I've been investigating reprogramming the Astras to say something other than "Virtual Office powered by Packet8" but when I try ADSIProg I get something about CPE

We have issues with all of the SPA-2000 ATAs we have where incoming calls from only one of our Asterisk servers do not complete. Details: 1- On the CLI we see that when the call is pushed to the ATA it shows Busy/Congested 2- We can make calls to this same server just fine 3- We can receive calls from other Asterisk servers running older CVS versions of Asterisk with the same exact ATA

Guys. Im having a big problem transfering incoming calls thru zap channels to some other extension. If the call is made by me to the outside via zap channels, no problem, hitting # gets me the transfer prompt, but if the call comes in thru zap and eventhough I am sending the call from the zap channel to my sip ata (GS ata 286) using Dial with wtWT as parameters, when hitting # I don't hear

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic

Hello, can anyone comment on how one could use SIPphone's $89 All-in-One adapter with Asterisk? Sounds to me like it should work as both a FXO and FXS. It would be a cheap way of getting started with Asterisk and PSTN. Any comments on the SIPphone FX200? Any comments on SIPphone in general? Thank you for your help

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

G'day all, I've been googling myself silly looking for help on this one but have come up blank. I have an AVM Fritz!Card PCI, and I'm using chan_capi v 0.3.1 with * from CVS-HEAD-05/08/04-22:48:00. I can start * and make and receive calls on ISDN fine but after a few hours of * uptime, on any ISDN call I make or receive from my SIP handsets (7960 or ATA-186) I get bad audio: on the

I have a lot of experience, all of it pretty good, with various Sipura products, Grandstreams, Zultys, IAXy, and numerous SIP/IAX soft phones connecting into Asterisk as clients. Good sound quality, great reliability. I've tried two of the units named in the subject line, and frankly I'm frustrated. Calls usually start out OK, but within a brief period the sound goes totally to

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

I'm having a horrible experience getting a Cisco ATA-186 to work with *. I can make calls from the ATA with no problems. However, incoming calls make the ATA ring once, and then the call is disconnected. I have no problems with my Sipura 2000 or my Grandstream phones. I am running 2.16.1 sip code on the ATA 186. Neither * nor the ATA is behind a NAT. They are both on public IP addresses

I have a SIP connection to Broadvoice and sometimes when I make outgoing calls from a SIP ATA-188 (could be the same number) (the ATA-188, is currently the only extension), there is no audio passed for 5-10 secs. I have set all the codec the same to 711u and also ensured canreinvite is set to no. Any suggestions? Places to look for? -------------- next part -------------- An HTML

How do I set asterisk not to answer incoming PSTN POTS calls? I want to be able to use the line for outgoing calls only. -Thanks Tim

http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com genom Brian J. Murrell Skickat: to 2006-02-02 20:14 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time I want Asterisk to delay answering the POTS line via a

I have an Asterisk 1.2.16 server running CentOS 4.4 with a TE110P card and an OpenVox A1200P card. Up to today everything was working perfectly. The OpenVox card has 8 FXS and 2 FXO ports. The two faxo ports are used for a GSM adapter and for an ATA connected to Vonage. The problem we started noticing today was that the Vonage line will receive a call and then cannot connect to any of the SIP