search for: gavinadam

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2006 Jan 27
0
Good provider of Polycom Phones (mostly for accessto latest/greatest firmware)
Stay away from Alliance Systems. We ordered $15k worth of Polycom's over a month ago and we're still waiting. Our account rep's communication with us on what the delay has been, has been terrible. Doug. -----Original Message----- From: Gavin Adams [mailto:me@gavinadams.org] Sent: Friday, January 27, 2006 8:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Good provider of Polycom Phones (mostly for accessto latest/greatest firmware) Hi, I've ordered a few IP501s from PC Connection, basically since we hav...
2006 Jan 27
1
Good provider of Polycom Phones (mostly for access to latest/greatest firmware)
...count with them. I like the phones for what they do, and now would like establish a relationship with a reseller that can give us maintenance and access to the most current firmware. What are some good resellers out there? Regards, --- Gavin Adams VP of Technology Promisant (USA) Inc. Email: me@gavinadams.org
2006 Mar 08
1
Location of MeetMe Recordings
...isk/asterisk.conf file is set to point to /var/spool/asterisk for recording related bits, and voicemail and general recordings 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: me@gavinadams.org Office: +1 678.281.6402 Fax: +1 678.281.6401 Mobile: +1 404.933.8183 Skype: gadams999
2006 Jan 21
1
Asterisk 1.2.2 - Double Quote on CallerID Causing SIP Problem (7940)
Hi, I just upgraded by 1.0.x home server to 1.2.2. Overall the upgrade went fine, but a strange problem has cropped up with the CALLERID name value of incoming calls from the X101P card. When an incoming call is presented (via Vonage ATA), the calledid value getting double quotes up from: -- Executing NoOp("Zap/1-1", """WIRELESS CALLE" <1404xxxxxxx>") in
2006 Mar 03
1
SIP Problem - Asterisk to Provider Gateway
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 takes place and the caller can hear about 1-2 seconds of audio before the SIP provider