similar to: No ringing on inbound DID calls

Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)

Greetings, I have a new * system installed and everything works as it should except for one annoying little problem: I can't play any sound files. What this means is that when an extension script gets to the point where it should play a sound file (voicemail greeting, auto-attendant, whatever), the caller hears a click and then silence. According to the * log, the sound file is being

If you have used * to support a pri as pri_net (as opposed to pri_cpe), either to talk to another * system or a PBX of some sort, I would be very interested in hearing about your experiences. Imparticular, I would like to know that it works before I invest in the extra hardware. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115

I am trying to figure out why calls between SIP devices and the PSTN are being regularly dropped after anywhere from 2-15 minutes. I have turned on everything I can think of, but I don't see any obvious reasons for the drops. All I can see from turning on debug and verbosity is two messages advising of a destroyed call, followed by normal-looking SIP and ZAP termination messages. The first

Asterisk apparently has five echo cancellation algorithms: STEVE, STEVE2, MARK, MARK2 and MARK3. The current default appears to be MARK2. My question is, has anyone had any experience with any of the others (other than MARK2), and is there some conventional wisdom as to when to use one over another? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? TIA Bruce Komito High Sierra Networks, Inc.

If you have any experience using * (or VoIP in general) with DirecWay, please respond privately. I am particularly interested in experiences in Latin America. TIA! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I am routing outgoing calls through a sip gateway. The calls go through no problem, however the ringing in the callers ear begins as soon as the last digit is dialed. This has two nasty side effects. First, the caller hears 1-2 more rings than the callee. Second, and more importantly, if the callee's line is busy, the caller continues to get hear ringing, even though the gateway has

Should be mailbox = 7752365815@vpbx-wpti Best Regards, Ben Bawkon --------- Original Message --------- From: Bruce Komito To: <asterisk-users@lists.digium.com> Subject: [Asterisk-Users] voicemail notify problem on sip extension Sent: 5/19/2004 4:27:51 PM I'm having a problem with the voicemail notify feature. Although I have the voicemail box configured for the sip extension, the

When I enable NOTIFY messages in my SIP device (Sipura), Asterisk reports: handle_request: Unknown SIP command 'NOTIFY' from 'xxx.xxx.xxx.xxx' When I disable NOTIFY messages, * reports the device UNREACHABLE, followed by REACHABLE every couple of minutes. I think I want NOTIFY on, because the Sipura is behind a NAT server, but the constant stream of warnings from * make me think

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I pick up the ringing phone, the call is uncerimoniously dumped. I can tell the call

Hi all, I bought a couple phones for really cheap just for a simple solution. I'm trying to get a few 7910 to work with *. I'm just not sure how to get them to work. The 7910 just sits there "configuring IP" Here is a copy of my skinny.conf. the extensions.conf is default. I just want to bring the system up in default before a start making changes. Do I need to make

When a user has voicemail, I would like * to call the user at a pre-determined number (internal or external) and play a message that the user has voicemail, and then give the user the option to login to voicemail and pick up the message. I know about the externnotify feature, but I don't see a way to use it to accomplish what I want. I've checked the archives, etc., but I don't see

I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2 of the card and is set as the primary timing source. It is ESF/B8ZS. All the software is latest

Greetings, I tried to build astcc, but the Makefile is looking for Asterisk/AGI.pm. Anyone have any idea where this file is supposed to be and where it comes from? I've dragged in everything I can think of from cvs, and * is otherwise running fine. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I've been using realtime to store my voicemail configuration in a mysql table for several months now, and have had no problems...until today. A few weeks ago, I upgraded to the latest CVS and today I noticed voicemail is not updating the password when the user changes it through option 0. I'm not sure when this started happening, but I assume it was sometime after I upgraded. Has anyone

Since connecting a PRI to a Digium T100P, I have been seeing the following messages in syslog every few minutes: Jun 4 06:51:54 pbx asterisk[13435]: WARNING[1209214400]: chan_zap.c:6176 in zt_pri_error: PRI: Read on 56 failed: Unknown error 500 Jun 4 06:51:54 pbx asterisk[13435]: NOTICE[1209214400]: chan_zap.c:6913 in pri_dchannel: PRI got event: 8 on span 1 Sometimes, these messages come out

I've got someone sat in a home-office with an SPA921 behind NAT, and most probably a firewall. I've got a STUN-server running, and calling in from the SPA921 to our Asterisk box works fine - though I had to open the firewall for UDP traffic on port 10000-20000. Calling from our Asterisk to the SPA921 doesn't work. I'm guessing this is due to the NAT/firewall on the other side,

I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a Linksys firewall that supports UPnP. The Asterisk server has a public IP. Here are the problems that I am having with this configuration... 1. The 2 SIP phones can call MeetMe and have a conference but cannot call each other. (Yes, they connect but no audio either direction) 2. I have verify=yes in the sip.conf for both

Enhanced OS. General rules I use: -Do not use SIP transformations (the VOIP tab), these cause random RTP = issues, and once you start forwarding calls between users, all things go = to heck. You are better off using NAT/qualify in your sip.conf. -Do not use SonicOS Standard (all new Sonicwalls should come with = Enhanced now anyway) as there is no method to increase the timeout for = UDP rules,