similar to: Cisco 79xx firmware

Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call

Hi, I've got some weird sound artifacts happening during calls, they're very hard to describe, so I have a 122kb recording: http://openprojects.rarcoa.com/~miztic/artifact.wav normally the artifacts are just short blips, not quite as long as the one above, but they sound the same. When using the aggressive echo suppressor, it seems like those artifacts cause a really loud buzzing sound to

Hi list, I was wondering if anyone had any solution to either one of two issues I'm having: I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware, it works very well for the most part, but after less then a week of heavy usage, eventually the phone gets into a state where it won't accept or let you place any more calls, the screen flashes "no free lines

When a call is placed out the Zap interface there is a long pause followed by an error message from the telco that the call can not be placed as dialed. We have a tdm2413e with 11 1FB (POTS) lines. The number being dialed is a working local number, all dialed numbers get the same error. What should we be looking for ? Best regards, Duane Pudenz Network Infrastructure Manager Shasta

Hi list, This is not necessarily an asterisk issue, but a lot of you guys know way more then me, so I have a question: someone at my company sets his phone to forward calls to his cellphone, so someone calls our office, call is forwarded to his cell, and the callerID that shows up on his cell is of course our office number, because asterisk originates a new call to his cell and then bridges

Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio

Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a

Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter > > Message: 5 > Date: Fri, 27 Aug 2004 08:45:19 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion >

Interesting story. I was following up any wild goose chase i could to find a resolution including cleaning dust out of the machine. At 4 pm I plugged the line in and called for the sake of luck, and it was back to working. In the mean time, thanks for the troubleshooting advice. I will keep this in my log book so I have the steps for next time something goes awry. ------------------------------

Hi everyone, I'm having a bit of an issue after upgrading from asterisk ~1.2.24 to 1.6.2.11, with the old version when the user would go to transfer a call, they would press Transfer, then the speed dial button for the extension, optionally introduce the call, and then press Transfer again to complete the transfer. Now, with the new version, when you hit the speed dial button, asterisk

The CDW part number is: 672205 And the cisco part number is: CON-SNT-CP7960 Hope this helps... In fact I think I will add it to the Wiki. ~c ------------------------------ Message: 11 Date: Mon, 27 Sep 2004 10:40:38 -0500 From: "W. Kevin Hunt" <Kevin@hbcorporate.com> Subject: RE: Cisco Downloads --> was --> Re: [Asterisk-Users] Cisco 7960 andAsterisk...not working... To:

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Paolo Bonzini <pbonzini at redhat.com> wrote: > Il 29/05/2014 19:45, Chris Webb ha scritto: >> Chris Webb <chris at arachsys.com> wrote: >> >>> My CPU flags inside the crashing guest look like this: >>> >>> fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush >>> mmx fxsr sse sse2 ht syscall nx mmxext

Paolo Bonzini <pbonzini at redhat.com> wrote: > Il 29/05/2014 19:45, Chris Webb ha scritto: >> Chris Webb <chris at arachsys.com> wrote: >> >>> My CPU flags inside the crashing guest look like this: >>> >>> fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush >>> mmx fxsr sse sse2 ht syscall nx mmxext

can you please share the cdw part # for the $ 10 service contract ? W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Christopher Jacob > Sent: Saturday, September 25, 2004 9:51 PM > To:

I know this is slightly OT but folks discuss zfs compatible hardware here all the time. :) Has anyone used something like this combination? <http://www.cdw.com/shop/products/default.aspx?EDC=1346664> <http://www.cdw.com/shop/products/default.aspx?EDC=1854700> It''d be nice to have externally accessible CF slots for my NAS. I can''t put them into a drive bay because

I just had to deal with this yesterday. I called Cisco and they gave me a part number for the support contract. I looked around and it was $90... I posted back to this list and was happy when someone gave me the correct part number, which at CDW was $10... Not too bad. Although I can't believe Cisco waists time with $10 service contracts. At that point just make the damn thing free. Anyway,

Hi, I have followed the thread on where to get a maintenance contract so that I can upgrade the firmware to a SIP version and have called two different places, CDW and Insight. Neither place will sell the contact, all I get is a B.S. excuse to why they can't. Has anyone had luck with a more friendly place or do they have the images? TIA, Trevor

I have a system in place where it appears that TCP will make a massive change in behavior mid-stream with existing SSH sessions. We noticed the issue first with an application using an SSH forward. However, we were able to rule that out by generating the same TCP characteristics by having a perl script dump text out to a terminal simulating a large data flow from the far end(ssh server) back

There is hotel application weary popular in Croatia - Micros-Fidelio. Now I need to connect Asterisk with this application for purpose of billing. Thing is that hotel would like to give customer one bill for every service that he used while he was in hotel. Has anybody connected Asterisk with Micros-Fidelio? As I understand this isn't some local developed application, it's something