[asterisk-users] Verizon ISDN service in NY Hunt Groups

group, I find that EXTREMELY hard to believe. If you've had success could you share the person you spoke with and or what you asked for (or better yet a tarriff #) thanks! Bernie ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

[Asterisk-Users] no ring to callers?

a call is recieved. what am i missing here? thanks Bernie -- Executing GotoIf(SIP/2017540019-fe29, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(SIP/2017540019-fe29, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn

Re: [Asterisk-Users] no ring to callers?

+440/100,!0/100,!350+440/100,!0/100,350+440 record = 1400/500,0/15000 info = !950/330,!1400/330,!1800/330,0 Eric Wieling aka ManxPower wrote: Bernie Courtney wrote: OK, i've got asterisk @ home 1.3 up and running with Broadvoice. BUT I have one nagging problem to sort out. When you call my BV

Re: [Asterisk-Users] no ring to callers?

no it wasnt there, but i tried adding it and it still didnt make a difference, still no ring :( Bernie Paul Belanger wrote: check in modules.conf: load=res_indications.so is it there? Bernie Courtney wrote: indications.conf reads as follows [general] country=us [us] description

[Asterisk-Users] Best VoIP provider

looking at setting up an asterisk box at my home-- what VOIP providers are you all using with the best results (and low costs! lol) thanks Bernie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo

[Asterisk-Users] configuring Asterisk and broadvoice

does anyboy have a how-to online on how to do this? so far i've got the following in my sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external

Re: [Asterisk-Users] configuring Asterisk and broadvoice

PROTECTED]' of Request 107: Found Jul 19 23:14:39 NOTICE[2364]: Failed to authenticate on REGISTER to ';tag=as11b598b3' Jul 19 23:14:39 NOTICE[2364]: Failed to authenticate on REGISTER to ';tag=as11b598b3' Patrick wrote: On Tue, 2005-07-19 at 13:37 -0400, Bernie Courtney wrote: [snip] port

Re: [Asterisk-Users] configuring Asterisk and broadvoice

Jul 19 23:38:06 VERBOSE[1191]: -- Got SIP response 404 Not Found back from 147.135.0.128 Bernie Courtney wrote: does anyboy have a how-to online on how to do this? so far i've got the following in my sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0

[Asterisk-Users] dialling in from analog line - only get 2 of 3 digits extensions

can be done to make asterisk behave nicely even here? Any info pointers are greatly appreciated. Honestly :) -- best, Bernie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: asterisk strips off trailing digit from incoming calls

! Bernie -- best, Bernie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Re: asterisk strips off trailing digit from incoming calls

Hi Ronald, * chopping after 10 digits is fine - our number is 12345673 digit ext though so there's a total of 9 digits. On 7/4/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote: Bernie Ott wrote: There's a tiny bit of new info available: asterisk only strips off the trailing digit of calls

[Asterisk-Users] annoying static when calling from legacy PBX - * ZAP interface

, nothing else. Now - what can be the issue here? Any help and pointers are very appreciated! -- best, Bernie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] cant connect

on? or anybody have an idea as to why its not working? thanks SEND TIME: 19038812 SEND 170.20.41.165:5060 REGISTER sip:170.20.41.165 SIP/2.0 Via: SIP/2.0/UDP 170.20.11.59:5060;rport;branch=z9hG4bKD170D188076045BE8656D59545FF9678 From: Bernie sip:[EMAIL PROTECTED];tag=2472129214 To: Bernie sip:[EMAIL

[Asterisk-Users] Asterisk @ Home AMP problem

I just installed the latest version .8 of * @ home and it looks like it installed successfully, however when i goto the box's IP address and click on link to access the management portal the default password (admin/password) do not work...anyone else have this issue? thanks bernie

[Asterisk-Users] bandwidth

how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just trying to plan out a system and need some figures to plan on bandwidth allocation. B ___ Asterisk-Users mailing list

Re: [Asterisk-Users] bandwidth

PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernie Sent: Monday, April 04, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] bandwidth how much bandwith is used to go between a phone set and the asterisk server when a call is in progress? Just

Re: [Asterisk-Users] SIP -h.323

-to-end or always via Asterisk? Another question I'd be interested in: Have you also gained some experience with bridging _video_ calls between H.323 and SIP? cheers, Bernie PS: I'd be glad, if I also could get the relevant config files from you. On Fri, 13 Aug 2004, Ryan Wilkins wrote: Yes

[Asterisk-Users] Video Calls between SIP and H.323

and the option videosupport=yes ? Did I forget anything in the config? Looking forward to your answers! cheers, Bernie PS: I am using * version 1.0-1 0, which is in the current default in the Debian Testing distribution. I am using chan_h323 (the * built in), (but not chan_oh323

[Asterisk-Users] Hwo to get CallerID: SIP - ISDN

, I guess. Does anybody have some experience in this? Any hints, instructions and HowTo's are warmly welcome. cheers, Bernie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE

[Asterisk-Users] Segmentation fault, exit status 139, ...

with XXX). Any idea what there could be the reason for this segmentaion fault? What other indormation (e.g. configs) would be required to analyse this problem further? Thanx for you help! cheers, Bernie * Console: Apr 5 18:01:18 WARNING[24594]: app_dial.c:331 wait_for_answer: Unable to forward