[Asterisk-Users] Help. New SIP hardphone.

I have an Avaya 4602SW SIP phone. They just released the SIP firmware for it the other day. I have it working with my Asterisk, but have a couple issues. My setup is like this: Avaya 4602 phone at home behind router and Asterisk server is straight on the Internet. My phone registers with

RE: [Asterisk-Users] Sipura-SPA2000 background noise

Not I. -Original Message- From: Kevin [mailto:[EMAIL PROTECTED] Sent: Tuesday, June 01, 2004 7:44 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sipura-SPA2000 background noise I have been using Cisco ATA's for analog connections and decided to give a Sipura SPA-2000 a try. I

RE: [Asterisk-Users] Sipura-spa2000

Are any of you guys able to use the Sipura-spa2000 with call waiting on a zap call? -Original Message- From: Kevin Walsh [mailto:[EMAIL PROTECTED] Sent: Monday, May 31, 2004 10:23 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sipura-spa2000 Simon Chappell [EMAIL PROTECTED]

RE: [Asterisk-Users] intercept ringing phone

I havent used it with Asterisk, but I think I have seen the feature. On my Definity, the feature is called Call Pickup and you add extensions to the call pickup group. -Original Message- From: Michael George [mailto:[EMAIL PROTECTED] Sent: Wed 5/26/2004 3:33 PM

[Asterisk-Users] dialout=fromvm

If you set dialout=fromvm in your voicemail.conf, how do you then go about being able to dial back out? Is there a service feature code? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To

RE: [Asterisk-Users] dialout=fromvm

login to the voicemail I hit '3' for Advanced Options, 4 is never an option and if I hit '4' anyway it just says 'you have no more messages' Suggestions? -Original Message- From: Boater Sent: Tuesday, May 25, 2004 9:24 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] dialout=fromvm

RE: [Asterisk-Users] How can I dial (0 + telephone number)

This is what I use. I too get dial tone to my fxo from another pbx which requires a '9' so when you look at my config, remember that I dial a 9 to get out. [zap_outgoing];LOCAL CALLING STARTexten = _XX.,1,Dial(Zap/1/9${EXTEN}#,20,r)exten = _XX.,2,Hangup;LOCAL CALLING END ;L/D

[Asterisk-Users] *, Sipura, Call-Waiting, X100P, 2 ZAP Calls

Does call-waiting work for anyone that recieves 2 pstn calls on a X100P using a Sipura? I have modified the dialplan in the Sipura such that the *0 is definately getting sent to the Asterisk server now. When the phone beeps and I flash hook I get tone, then dial *0# and the sip debug shows