Re: [Asterisk-Users] sip change?

2004-08-27 Thread Doug Shubert
has anyone tested the vst1000 SIP phone from pcphoneline ? http://www.pcphoneline.com/ Doug Jerry Roy wrote: Hi All, Looking for a recommendation. I was hoping to purchase a * "KIT" for a small office. I have 4 lines and 4 extensions need phones so I need 4 phones. What phones would many

RE: [Asterisk-Users] sip change?

2004-08-27 Thread Chad Brown
, August 27, 2004 10:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] sip change? Hi All, Looking for a recommendation. I was hoping to purchase a * "KIT" for a small office. I have 4 lines and 4 extensions need phones so I need 4 phones. What ph

RE: [Asterisk-Users] sip change?

2004-08-27 Thread Jerry Roy
Hi All, Looking for a recommendation. I was hoping to purchase a * "KIT" for a small office. I have 4 lines and 4 extensions need phones so I need 4 phones. What phones would many of you recommend? Can you refer me to any companies that have built a kit I can plugin and configure? Thanks, Jerry

Re: [Asterisk-Users] sip change?

2004-08-27 Thread Rich Adamson
* and the 7960's are on the same wire, no firewall involved whatsoever. Backing out to July 12th now... > Whenever I see the "Maximum retries" message it usually indicated a > communication problem, like one way traffic. Last time I got it, I > traced it to a bad firewa

Re: [Asterisk-Users] sip change?

2004-08-27 Thread Deon Rodden
Whenever I see the "Maximum retries" message it usually indicated a communication problem, like one way traffic. Last time I got it, I traced it to a bad firewall rule, dropped the firewall and it worked, the time before that when I received it, it was due to a routing error, the server could g

OT re: [Asterisk-Users] sip change?

2004-08-27 Thread Matt Schulte
Kind of off topic but I know CVS is the "prefered" way of upgrading, however are there such things as "stable" CVS upgrades? It seems a lot of the CVS's have a lot of devel bugs in this that I would be scared to put even near production. Just IMHO. :-) Matt -Original Message- From

[Asterisk-Dev] Re: [Asterisk-Users] SIP change...

2003-08-24 Thread Dave Packham
Thanks for the reply. snips of it are in the Cisco TAC case logs and developers are looking at it. Ill let you know if I get a resoloution Dave P >>> [EMAIL PROTECTED] 8/23/2003 12:53:11 PM >>> > Normally the caller-id is taken from "remote-party-id" in the SIP > INVITE. We don't see that fie

Re: [Asterisk-Users] SIP change...

2003-08-23 Thread Mark Spencer
> Normally the caller-id is taken from "remote-party-id" in the SIP > INVITE. We don't see that field poplated in this INVITE. What is the > originating gateway? What device is sending the call to the 827? We > should be seeing "remote-party-id" in the INVITE. The string "remote-party-id" does

Re: [Asterisk-Users] SIP change...

2003-08-23 Thread Dave Packham
Interesting. I am working on getting CID to work from * to my Cisco routers. I have a tac case open and they are giving me debug IOS's to work with but this is what they have come up with. Dont know if this will help quoted from my talks with Cisco TAC Hi Dave - A few more questions from