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...NBOX.wav" Another little observation is that a few of the samples sound a bit "clicky", like they don't start cleanly on a zero crossing... and a couple have variable volume/level... have you put them through something like Audacity and cleaned them up? Regards Mike Tubby G8TIC PS. We do IVR recordings too ... you can call our "virtual switchboard" on: 01905 756700 (UK) or +44 1905 756 700 (Intl) where you'll hear my friend Heidi ... or 845-613 1328 if you're in the USA! ----- Original Message ----- From: "Peter Bowyer" <...

...ority 10 PABX ! emergency calls 911/999 gw-type-prefix 1#* default-technology no shutdown ! ... so I think this should route the calls between the gateways...? Mike ----- Original Message ----- From: "Stephen Bosch" <posting@vodacomm.ca> To: "Michael J. Tubby G8TIC" <mike.tubby@thorcom.co.uk> Sent: Monday, February 05, 2007 7:51 PM Subject: Re: [asterisk-users] Help sought: Asterisk H.323, Cisco IOS Gatekeeper(s) intra-zone call routing and TETRA > Michael J. Tubby G8TIC wrote: >> I can see how this be acheived if I had two Gatekeepers...

I am looking at phones for my asterisk system and seem to have a problem. The only Power over Ethernet phones I can find that support the IEEE standard are 3com. Cisco uses its own proprietary ( and is expensive to boot ), snom has a different but equally non-IEEE method, and i'm havent found another phone that I'm confident can do the job for our office. Whats a good high quality ip

...vailable. Again, thanks to all who are offering their assistance, I look forwards to hearing from you, and all being well meeting you soon. Cheers, Ben -----Original Message----- From: asterisk-uk-bounces@xdev.net [mailto:asterisk-uk-bounces@xdev.net] On Behalf Of Michael J. Tubby B.Sc (Hons) G8TIC Sent: 22 March 2005 15:31 To: Asterisk UK Community mailing list Cc: Simon Clifton Subject: Re: [Asterisk-uk] Meet ----- Original Message ----- From: "Peter Bowyer" <peeebeee@gmail.com> To: "Asterisk UK Community mailing list" <asterisk-uk@xdev.net> Sent: Tuesday,...

Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001>

All, I have upgraded by home machine from Fedora Core 5 (FC5) to the recent FC6 and am struggling to build Zaptel-1.2.10 and Asterisk-1.2.13 on the box... which is an Intep P4 2.8GHz HT processor box with 845 chipset, hence the kernel installed is 2.6.18-1.2798.fc6-i686 so we hve this: [root@gate zaptel-1.2.10]# uname -a Linux gate.tubby.org 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:54:20

All, I've fixed my problem getting Asterisk ooh323 channel to stay registered with my Cisco IOPS gatekeeper, now I need to get dialling working. I have the following: [Asterisk with ooh323] ----h323---- [Cisco IOS GK] ----h323---- [Radio system OpenH323] 192.168.1.5 192.168.1.6 192.168.1.7 the Asterisk box has numbers

All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form

All, I'm running (attempting to) ooh323 with Asterisk and a Cisco 2621XM router operating as a H.323 GateKeeper, however when I bring the Asterisk box up it registers successfully with the GateKeeper (exchanges GRQ/GCF, then RRQ/RCF) it notes the GateKeeper supports keepalive at 300 seconds, when it gets to time to re-register its sends an RRQ again and gets rejected with RRJ (unspecified