[asterisk-users] Avaya IPOffice and a SIP trunk to Asterisk

Has anyone successfully setup the Avaya IPOffice 500 with a sip trunk to Asterisk. If so can someone give some config examples? Thanks Rick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list

RE: [asterisk-users] multiple phones registered for the same user

Create a different user for each phone and create a ring group with the phones that you want to ring. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Wednesday, February 28, 2007 9:15 AM To: Asterisk Users Mailing List -

RE: [asterisk-users] TE110P: Error == Asterisk died with code 1.

I see that you have signaling listed twice. That might be causing a problem. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, February 27, 2007 6:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

[asterisk-users] Need info for creating * as a gateway for other * servers.

Can someone point me to some documentation on how to configure an Asterisk box to do Termination and Origination for a few other Asterisk servers? We have a box with a T-1 in it and we want to share it with some other companies that have Asterisk servers via SIP and or IAX. In particular we

RE: [asterisk-users] Polycom 501 One-way Audio

Had the same issue and it was the handset cord. Rick From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Thursday, September 28, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 501

RE: [asterisk-users] problems with Polycom 500 boot up

I am not positive but I thought that the 2.6.2 bootrom was the highest you could put on the ip500. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jessee J Holmes Sent: Wednesday, September 20, 2006 2:41 PM To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] SNMP with 1.2.11 stable

Can anyone tell me if it is possible to get the asterisk SNMP module working with ver 1.2.11 stable. Everything that I am coming across is talking about using the trunk version. If it is how do I get it compiled with this version? Thanks Rick

[Asterisk-Users] update or add DID's to directory Assistance

I need some help getting our DIDs updated in the directory assistance and also the caller ID cname that is displayed. Does anyone know where to go to do this so the major carriers will get this info? I have found www.listyourself.net but I am hesitating to submit info to someone I have

RE: [Asterisk-Users] Re: How is Teliax ?

I would not ride on a tracert too much. We use Teliax also and our ISP that we have at the data center switched there backbones around the same time Teliax where doing there upgrades. We started seeing some call issues and when we did a tracert we started getting some dropped tracert responses

RE: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3

We had the same issue but we found that it was really the MS proxy server that the phone was going though. Set it up to use a different route out to the server and everything worked fine. Had to prove it to the admin at the location too, that was fun! Rick -Original Message- From:

RE: [Asterisk-Users] Polycom 501

Try and download the correct sip.cfg for your boot ROM ver from here and see if it corrects the problem. We use AMP with these files and we never had an issue with the transfer button not working. http://www.freedomphones.net/polycom/files/ Make sure that you reboot the phone after

RE: [Asterisk-Users] Creating conf files from db

AMP does this. http://coalescentsystems.ca/ Rick From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, December 22, 2005 3:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Creating conf