similar to: IAX2: Connection rejected

Hi there, Does someone can share the experience with Cisco and Polycom Phones? How rock solid are they? And who will win in sound quality contest? I heard that Cisco phones is a Polycom replicas with changed design. Is that true? What else phones is better to implement to the medium sized business? The rock solid stability and superb sound quality is a must. -- All the Best! Sergey.

Hi All, The wiki seems to be struggling this evening. Anyone else seeing this? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

Hi, Have you tried it? Any comments would be greatly appreciated. I can have it at C$200, is that a good price? Thanks a lot. robert

Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/8cdbeeec/attachment.htm

Hi all. I am going to do a simple "voting application" for a radiostation. The idea is to have listeners call in to vote on songs. What I want to do is to take a phonenumer for each song and present the result on a simple webpage. Eg. To vote on song number one, call 555-1111 To vote on song number two, call 555-2222 etc etc. When the listener calls in, a playback tells him:

I've been doing a lot of background reading/searching of this list, voip-info.org, and Google, looking to define a good candidate for a server platform. I'm very interested in thoughts from others! So here goes... Axiom 1: if you are not doing doing much transcoding (converting between codecs), the bottleneck for supporting high volumes of simultaneous calls is system bus speed,

On one of my 3 * servers I get this after 2 or 3 IAX2 calls Apr 22 15:54:39 NOTICE[1150495040]: chan_iax2.c:1271 iax2_destroy: Avoiding IAX destroy deadlock And as if that wasn't enough I get a never ending stream of this error flying off the top of the screen. At which point I can no longer make any calls into or out of the box. Any commands issued at the CLI prompt are ignored so I have

Hi, cisco's phones are VoIP only polycom build (video-) conferencing devices. One Cisco model (7930 I thnk) is a polycom in disguise. The code is not 'cisco-like' (at least the version I had. Both brands make very good quality equipments. Good sound, good support, ... Regards, Shaoul Jacobson VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail

Anyone suffering an outage with them right now? I am getting the following from my box when I try to dial using them.... == No one is available to answer at this time W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/e1096325/attachment.htm

Hi Kostya, >On Wed, Nov 12, 2014 at 2:50 AM, Volodymyr Kuznetsov <vova.kuznetsov at epfl.ch >> wrote: > >> Dear LLVM developers, >> >> We've applied the feedback we received on Phabricator on the SafeStack >> patches, >> > >Did you investigate the possibility of moving the transformation from >codegen to the LLVM level, i.e. the same level

https://bugs.freedesktop.org/show_bug.cgi?id=110660 Bug ID: 110660 Summary: GeForce GT 750M Mac Edition fullscreen issues Product: Mesa Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Drivers/DRI/nouveau Assignee: nouveau

Dear list i am using Asterisk 1.2.5 with A@H . here is my problem. if i dial a number (consider 79) i have to wait around 20 seconds before my Asteisk box response. now i want to decrease this waiting time . any idea how to do that ? thanks Salaque

+nlewycky On Mon, Nov 17, 2014 at 9:36 AM, Volodymyr Kuznetsov <vova.kuznetsov at epfl.ch > wrote: > Hi Kostya, > > On Sat, Nov 15, 2014 at 1:53 PM, Volodymyr Kuznetsov < > vova.kuznetsov at epfl.ch> wrote: > > Do you think moving the pass to lib/Transform/Instrumentation but > > scheduling it during code generation would make sense ? If so, we'll >

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

On Thu, 9 Feb 2017, Vitaly Kuznetsov wrote: > +#ifdef CONFIG_HYPERV_TSCPAGE > +static notrace u64 vread_hvclock(int *mode) > +{ > + const struct ms_hyperv_tsc_page *tsc_pg = > + (const struct ms_hyperv_tsc_page *)&hvclock_page; > + u64 sequence, scale, offset, current_tick, cur_tsc; > + > + while (1) { > + sequence = READ_ONCE(tsc_pg->tsc_sequence); > + if

On Thu, 9 Feb 2017, Vitaly Kuznetsov wrote: > +#ifdef CONFIG_HYPERV_TSCPAGE > +static notrace u64 vread_hvclock(int *mode) > +{ > + const struct ms_hyperv_tsc_page *tsc_pg = > + (const struct ms_hyperv_tsc_page *)&hvclock_page; > + u64 sequence, scale, offset, current_tick, cur_tsc; > + > + while (1) { > + sequence = READ_ONCE(tsc_pg->tsc_sequence); > + if

Hey, For the bridge issue, take a look at 'notransfer=yes' option in your iax.conf. It'll force * to stay in the path http://www.mail-archive.com/asterisk-users@lists.digium.com/msg42262.html

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

https://bugs.freedesktop.org/show_bug.cgi?id=110973 Bug ID: 110973 Summary: GeForce GT 750M Mac Edition backlight Product: Mesa Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Drivers/DRI/nouveau Assignee: nouveau at

Hi folks, Just a quick question. Does the GrandStream GXP-2000 phone support presence (hints)? Cheers, Richard. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060109/26c4a63c/attachment.htm