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November? It's DD/MM/YYYY in his case, not MM/DD/YYYY. Either way, even two days is more than enough for me. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of john beaman Sent: Tuesday, April 03, 2007 12:43 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: asterisk-users Digest, Vol 33,

Look at app_valetparking here: http://www.voip-info.org/wiki/index.php?page=Asterisk+addons ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Kevin Kiely Sent: Monday, July 16, 2007 16:47 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MultiParking

Does anyone have any opinions on the music on hold quality over G729? The stock files seem to sound terrible over it, this is enhanced further by calls coming from the PSTN via a Zaptel gateway. I am only using the stock wav files and have not attempted to use much else so far. I've ruled out timing issues on the system generating the MOH itself (ztdummy on the PBX itself, our Zaptel gateway

Hi, We have a PRI connection & when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 & RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats

...ted by way of a router -- but not necessarily NAT'd? -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671 ------------------------------ Message: 5 Date: Tue, 21 Aug 2007 20:24:14 -0700 From: "Darryl Dunkin" <ddunkin at netos.net> Subject: Re: [asterisk-users] Polycom and NAT To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <EC5DC6B7B2274B47A28A30C5FCF8744001AF2C6D at exchange.nosi.netos.com> Content-Type: text/plain; chars...

Question was off topic for the thread, but I'm feeling helpful today. More of a 1234... make install modprobe usb-uhci modprobe zaptel modprobe ztdummy -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brad Sumrall Sent: Monday, March 19, 2007 13:17 To: 'Asterisk Users Mailing List - Non-Commercial

You have it right, for 1.2, use 'tos=', for 1.4 use 'tos_sip/tos_audio/tos_video'. ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Al lists Sent: Monday, July 23, 2007 10:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Fwd: Asterisk and COS

In your per-phone configuration: <phone1> <reg> ... <divert divert.fwd.1.enabled = "0" divert.fwd.2.enabled = "0" divert.fwd.3.enabled = "0" divert.fwd.4.enabled = "0" divert.fwd.5.enabled = "0" divert.fwd.6.enabled = "0" /> This removes the soft-key and disallows the

Is there a way to disable ringing while dialing? Example, external users come into our IVR, and if they dial certain IVR options, these are sent off to a remote server for call handling ( Dial(SIP/extension at remoteserver) for example). It rings once, then the remote system picks up. I would like it to be more transparent to the users.