RE: [Asterisk-Users] Native MOH - Convert mp3 to ulaw

2006-03-21 Thread Gavin Adams
> From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Douglas Garstang > > I tried that earlier today... found it somewhere online... This is what I > get... > > [EMAIL PROTECTED] mp3]# sox -V fpm-calm-river.mp3 -t au -r 8000 -U -b -c 1 > fpm-calm-river.ulaw resample

[Asterisk-Users] Location of MeetMe Recordings

2006-03-08 Thread Gavin Adams
are being stored in the appropriate subdirectories. It's only meetme that is going to a different place. Regards, --- Gavin Adams VP Operations PARC Inc. E-mail: [EMAIL PROTECTED] Office: +1 678.281.6402 Fax: +1 678.281.6401 Mobile: +1 404.933.8183 Skype: gada

Re: [Asterisk-Users] SIP Problem - Asterisk to Provider Gateway

2006-03-04 Thread Gavin Adams
On Mar 3, 2006, at 1:46 PM, Gavin Adams wrote: Hi All, I'm stumped on a weird problem. I have an * server working fine for local SIP phones and IAX2 connections. We just provisioned a second Ethernet port to attach to a local SIP provider. PSTN calls incoming work fine: PSTN -

[Asterisk-Users] SIP Problem - Asterisk to Provider Gateway

2006-03-03 Thread Gavin Adams
Hi All, I'm stumped on a weird problem. I have an * server working fine for local SIP phones and IAX2 connections. We just provisioned a second Ethernet port to attach to a local SIP provider. PSTN calls incoming work fine: PSTN -> SIP Provider -> SIP -> * but outgoing calls are not. Call setup

RE: [Asterisk-Users] Re: Polycom 501 horrible echo

2006-01-30 Thread Gavin Adams
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > Upgrading to the correct sip.cfg fixed the problem. The Polycoms are > back to their great speakerphone-ness. A gotcha is that the new > sip.cfg now contains ntp settings. You'll need to modify these to fit > your time

[Asterisk-Users] Good provider of Polycom Phones (mostly for access to latest/greatest firmware)

2006-01-27 Thread Gavin Adams
s out there? Regards, --- Gavin Adams VP of Technology Promisant (USA) Inc. Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digiu

RE: [Asterisk-Users] Polycom 501 horrible echo

2006-01-23 Thread Gavin Adams
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Jeff Herring > I have the following situation: > > Asterisk 1.2.1 > 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 > Some 501's local to my network, some across th

RE: [Asterisk-Users] Asterisk 1.2.2 - Double Quote on CallerIDCausingSIP Problem (7940)

2006-01-21 Thread Gavin Adams
ay, I have a quick fix in place by removing the offending " character, which makes this better. I'll hook up the line to the a regular POTs and see if there is a difference the way the ATA-186 presents CID vs. HellSouth. Regards, --- Gavin Adams VP of Technology Promisant (USA) Inc. Em

[Asterisk-Users] Asterisk 1.2.2 - Double Quote on CallerID Causing SIP Problem (7940)

2006-01-21 Thread Gavin Adams
callerid="My name" <404 xxx > channel=>2 I'm running this under CentOS 4.2, fully yumified. Not being familiar with the 1.2 syntax, should I be looking in zaptel or asterisk to figure this out? I'll try to rebuild and install zaptel. Maybe the extra quote i