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...the only way I know to contact the person concerned] This message is for Stephen Karrington - it appears that you have over-agressive 'spam' filters and we can no longer email you. Please rectify this if we are to have meaningful conversation! The original message was received from Linus Surguy <linus@magrathea-telecom.co.uk> ----- This message has been blocked by our spam filter. ----- ----- If this has been a mistake, please contact ----- ----- the recipient through other means. ----- Undelivered message for: <"Stephen Karrington" <...

...est..." Today's Topics: 1. Re: G.723 (Eric Wieling) 2. RE: G.723 (Steven Critchfield) 3. Re: Voicemail storage in DB (James H. Cloos Jr.) 4. Voicemail config from database (AJ Grinnell) 5. oob to inband dtmf over rtp (James H. Cloos Jr.) 6. OT appologies to list (Linus Surguy) 7. Re: OT appologies to list (Brian Cuthie) 8. Zapateller issues (Mark Phillips) 9. RE: Zapateller issues (Sean Cheesman) 10. RE: Zapateller issues (Darrin Johnson) 11. Re: Re: Voicemail storage in DB (James H. Thompson) 12. Re[2]: [Asterisk-Users] OT appologies to list (Stephen Kar...

HI all, Having a mental block today - can someone confirm which direction the echo cancelation applies to for the Zap PRI channels? ie. is it removing traces of the transmitted data to the PSTN from the received data, *or* is it removing traces of the data transmitted to Asterisk from the data received back from Asterisk? Got a configuration that is based on a call made from the PSTN to a SIP

DID's from Allstream (AT&T) are $2 Cdn/month but I think they have a rule that it has to terminate on their network somewhere... -----Original Message----- From: Linus Surguy [mailto:linus@magrathea-telecom.co.uk] Sent: Monday, June 14, 2004 6:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Canadian DID Can anyone point me in the direction of a wholesaler of Canadian DID numbers? If they'd be interested in trading them for UK numbering that wo...

Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only

Hello all, Last couple weeks we had a lot of business discussions on mailing list, however some people don't like it, some people don't needed it, etc. I had couple discussions with Asterisk community members, who is interested to have business discussions about Asterisk, including but not limited to : business implementations, reselling , Asterisk commercial packages, IP phones,

My wife's got an appropriate Southern England (Wimbledon) accent and I'm sure she would try her hand. Does anyone have a comprehensive list of the words that need to be said? Matt, do you have them if your wife's done a set for French users? Mark, if you have the kit maybe you could chop up the file? I write a utility to chop up and compress the wave file based on some of the C

We're in the process of testing some equipment and configurations and to do this we have setup a UK PSTN Gateway to Free World Dialup. Simply dial 0845 004 5566 (UK local rate call) and at the prompt enter the FWD subscriber number - within a couple of seconds you should be connected. We can also terminate UK 0800/0808 numbers for SIP/IAX -> PSTN calls, at the moment we don't have an

Assuming I've got a setup where calls entering Asterisk on SIP leave on IAX2 ( and the reverse), i.e. a SIP user might dial '1234' where we then have extern => 1234,1,Dial(IAX2/somewhereelse) Now, we don't have any G.729 functionality on this server, so what happens if the SIP user calls with G.729 only available? Assuming the remote IAX2 server does have G.729 can it be

Hi all, Is anyone here using linux as a router and managing their VoIP traffic with CBQ ? If so, do you have any configs (tc scripts etc) to share? We've trying to ensure that all VoIP traffic is prioritised ahead of 'normal' traffic, and at the moment have setup two classes based on the TOS flags. It all seems to work perfectly, but it would be interesting to see what other people

I know this is OT for this list, but I havnt seen it mentioned here and in the spirit of 'open source' I thought this would be interesting for readers here: ----- Original Message ----- From: "Jeff Pulver" <jeff@pulver.com> Sent: Tuesday, December 23, 2003 11:28 PM Subject: [FWD] FWD Holiday Promotion: Free Calling to 8 Countries > Hi There, > > In the spirit of

Hi all, We've got a number of users connected in a configuration which is basically: (a) SIP Phones -> Asterisk -> IAX -> Our Asterisk -> Cisco AS 5xxx (SIP) -> PSTN We also have users in a configuration: (b) SIP Phones -> Asterisk -> IAX -> Our Asterisk -> Digium E1 -> PSTN The second server in both cases is the same. The SIP phones place a call via these