Re: [asterisk-users] Experiences with grandstream GXW 4024 FXS?

I have 4 instalations with with it, the first one from agust, and witout incident, in this time we make only 1 restart. Regards Mehdi Chouikh http://www.voz-ip.info http://www.unitelexperts.com http://www.mitelefonovirtual.com On Wed, Dec 24, 2008 at 6:05 PM, Daniel Hazelbaker dan

Re: [asterisk-users] Asterisk access Postgres for Realtime Configuration

Yes you can use res_conf_pgsql.so is present in asterisk 1.4 On Oct 7, 2006 1:22 AM, John Miloo [EMAIL PROTECTED] wrote: Hello Comunity, How can I get Asterisk realtime working with Postgres? (without ODBC)? Thanks John /doc/realtime.txt in Version 1.4 Beta2 Currently there are

Re: [asterisk-users] 1.4 SIP Jitter Buffer

I have asterisk 1.4.14 in 3 of my 8 servers for 3 weeks on productions systems, but i had problem with adapative jitter buffer, when i active it there are no sound. Regards On Nov 6, 2007 9:16 PM, Gregory Boehnlein [EMAIL PROTECTED] wrote: Are you running the SIP Jitter Buffer?

Re: [asterisk-users] Problem with conferences, Vlada, Pancevo

the forst problem you have, you need to los the meetme module, and second one is a timer, for that you can use ztdummy, compiling the zaptel driver. Regards On 5/7/07, Ronaldo [EMAIL PROTECTED] wrote: Hi, I'm not sure, but MeetMe needs some timer module from zaptel project. Try read about

Re: [asterisk-users] Hardware Suggestion for 2 PRI with call recording

We have a similar system up and runing for 6 months, wiith 60 channels, and average of simultaneas recorded calls us between 20 and 30. We make test for recording 60 calls without any problems We use a PIV Dual core with 3.2 Ghz with 2 mb of cache and 1Gb Ram. regards Mehdi On 12/13/06, A.R.

Re: [asterisk-users] FAX using T38

Hello Asterisk implement only passtrough T.38, so you cant terminate calls with asterisk using T.38. You need T.38 gateways. Regards On 11/13/06, Ricardo Carvalho [EMAIL PROTECTED] wrote: Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the

[Asterisk-Users] Moscow Dids

Hello I need Moscow dids urgently, Contact me offline [EMAIL PROTECTED] Regards Mehdi Chouikh Universal Telecom Spain ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] looking for Russia and Israel Dids

hello I am looking for Israel and Russia DiDs. Please email me to [EMAIL PROTECTED] REgards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

Re: [Asterisk-Users] oh323 or h323

Hello Personaly i prefer oh323, i am using for one year whitout problems. and is more easier to configure. regards On 9/1/05, Steve Ducat [EMAIL PROTECTED] wrote: I have just signed up for 2 landline numbers in China. They haveoffered to sell me 2 h323 compatible handsets which I have declined

Re: [Asterisk-Users] app_dial.c:977 dial_exec_full: Unable to createchannel of type 'Zap' (cause 0)

The clone work good, Try to change your dialplan for zap channel and it will work. Try this exten =_11.,1,Dial(Zap/1/${EXTEN:2},90,Tt). It work for mi with my clone card and my x100p (original card) for long time. saludos - Original Message - From: Rich Adamson [EMAIL PROTECTED]

Re: [Asterisk-Users] Hardware Capacity/Configuration

If use Alaw or ulaw as codec, i think it's enough. But if you need to make transcoding to a hard codec like g729, g723, you have to look other cpu. regards - Original Message - From: Daniel Salama [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [Asterisk-Users] asterisk to analog pbx

Hello all is right, the analog extension should ring, but maybe your dialplan is not correct or you call a bad extension in you PBX. can you post your dialplan?, to see it. regards - Original Message - From: Julio Saura [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday,

[Asterisk-Users] A-Z Termination

Hello I am looking for A-Z termination please send me your prices off-line. Protocols: SIP, IAX Codecs: G723, G729, GSM Regards Mehdi Chouikh Universal Telecom www.unitelexperts.com Tel: +34 902023154 ___ Asterisk-Users mailing list Asterisk-Users