similar to: How often do YOU register?

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

Dear All : I need to link my HQ to some Remote Sites - I need a Router which supports VOIP , and VPN Also the Router Should has 3 FXS ports and 1 FXO ... The call should be routed from the Remote Site to the HQ through a VPN tunnel ( 3DES ) ... Any Advise ? Mohamed Farid ,, Notice: This e-mail (including attachments) is confidential and is intended solely for the addressee. Unless you are

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May

Does any one know if there's an mechanism (internal to asterisk or otherwise) to replicate dynamic SIP device registrations across a pool of asterisk servers? I'm in the process of creating a asterisk cluster using a SIP hardware load balancer and so far this is one of the challenges I'm facing. One thought I'm currently investigating is to use openSER to intercept and

Hi, I wanted to let you know that I have just made available the sources to the ogg + vorbis implementation in haXe, which I've been working on for last couple of weeks. The code compiles to an swf file playable in Flash Player 10. A demo of a simple player implementation (latest Flash 10 required): http://people.xiph.org/~arek/pg/hx/test.html and the sources, in a bzr branch, currently

Hi, I wanted to let you know that I have just made available the sources to the ogg + vorbis implementation in haXe, which I've been working on for last couple of weeks. The code compiles to an swf file playable in Flash Player 10. A demo of a simple player implementation (latest Flash 10 required): http://people.xiph.org/~arek/pg/hx/test.html and the sources, in a bzr branch, currently

Dear all, As the next version of XiphQT is mostly ready, I thought it could use some more wide pre-release testing. The major change since last release is the addition of Ogg exporter and Vorbis and Theora encoders. Any feedback on how this new functionality performs (or doesn't!) with audio/video editing/producing software will really help. Also, comments and suggestions on the work of

Dear all, As the next version of XiphQT is mostly ready, I thought it could use some more wide pre-release testing. The major change since last release is the addition of Ogg exporter and Vorbis and Theora encoders. Any feedback on how this new functionality performs (or doesn't!) with audio/video editing/producing software will really help. Also, comments and suggestions on the work of

Dear all, As the next version of XiphQT is mostly ready, I thought it could use some more wide pre-release testing. The major change since last release is the addition of Ogg exporter and Vorbis and Theora encoders. Any feedback on how this new functionality performs (or doesn't!) with audio/video editing/producing software will really help. Also, comments and suggestions on the work of

Hi, I've just built fresh binaries of XiphQT - with trunks of Xiph libs and FLAC from the end of the last year. Code-wise there are few changes in the components since the last release: a memory leak fixed and the recently reported issue with iMovie'08 solved. I'd appreciate any help testing, especially on PPC as I don't have access to that architecture anymore. You can find the

Hi, I've just built fresh binaries of XiphQT - with trunks of Xiph libs and FLAC from the end of the last year. Code-wise there are few changes in the components since the last release: a memory leak fixed and the recently reported issue with iMovie'08 solved. I'd appreciate any help testing, especially on PPC as I don't have access to that architecture anymore. You can find the

Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.

On Jan 2, 2007, at 5:15 AM, Arek Korbik wrote: > The XCode project files you found are meant to be used with FLAC > 1.1.2. The FLAC repository now contains version 1.1.3 files, and there > have been interface changes in that latest revision > (http://flac.sourceforge.net/changelog.html#flac_1_1_3). That could > explain your problems with compilation. Well, now, no, I did download

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations?

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

HI All, I am using this code here: setTimeout("Sound.play(''sounds/movieprojector.mp3'');", 600); setTimeout("Effect.Appear(''slideshow5'');", 850 + 600); setTimeout("Effect.Puff(''slideshow5'');", 850 + 600 + 850);

Hi, I just purchased a 9331i phone. It's great! However, having a slight issue with line 2 and 3. line 1 works fine. But when I try to make a call with line 2 or 3 I get a fast busy from the phone. Asterisk shows this: Mar 23 15:19:12 NOTICE[18185] chan_sip.c: Failed to authenticate user No User <sip:No%20User@mysipserver.myhost.com:5060>;tag=34c1cd2bf55cb8d Can anyone with a

Arek, This is awesome information. Why not chuck it on the web page at http://xiph.org/quicktime/development.html ? Cheers, Silvia. On Mon, Aug 23, 2010 at 7:07 PM, Arek Korbik <arkadini at gmail.com> wrote: > Hi, > > On Sun, Aug 22, 2010 at 10:28 PM, G S <stokestack at gmail.com> wrote: > > Hi all. > > > > Maybe there's a document about the build

Hi Everyone: I have a THREE questions. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server and OR a registration server. Second question, with Asterisk are you able to do video on VOIP video phones? Last question, does

Hi, I just read thru the changelog.txt of the current CVS version and what catched my eye was the following line: 'Adding Q.SIG switchtype option to chan_zap' . But there is no sample config in zapata.conf for Q.SIG and no 'feature-list'. Does this exist anywhere or has anyone already has experience with * and Q.SIG and wants to share ?? Thanks a lot in advance, best