similar to: Meetme delay?

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)

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

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

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.

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

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

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

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,

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 was very pleased to learn that 1.4.5 has been released. Unfortunately, I have been beating my head against a wall trying to install 1.4.4 on FreeBSD (6.2). If you have been successful in building 1.4.anything (including addons and zaptel-bsd-trunk), could you please respond, on- or off-list with the secret? Plain vanilla asterisk is ok (I think), but when I try to build asterisk-addons-1.4.1

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

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

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 have several SIP devices (Sipuras) that are working fine with *, except for one annoying little problem. Occassionally, after being registered for some period of time, the Sipura returns a 404 Not Found to (I assume) an INVITE request. Of course, this makes the extension appear busy. When this happens, I check the Sipura and it is thinks it is still registered and I check * and it shows

If you have a PRI terminated in a Cisco router talking SIP to * and would be willing to share your Cisco config, please respond. Also, I would be interested in knowing what version of IOS you are using. We are using an NM-HDV in a 3640. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

I'm trying to get a ZIP 4x4 working behind a NAT server, talking to * on a public address. When I use the same sip.conf configuration (and same NAT server) that works for Grandstream and Sipura phones, the 4x4 can register and make calls, calls *to* the 4x4 do not make it to the phone. I can see from the sip trace that the sip packets to the phone are being retried by *, but I don't

I would like to configure my Zip2 phones via tftp, however the tokens in the config file are (apparently) not all documented. Specifically, the username/password/callerid fields seem to be only configurable via the web interface. I find this hard to believe, but the documentation and examples that Zultys provides don't help. If you have an example of a tftp-loadable config file for the Zip2

I installed the mysql/voicemail addon, and it works very nicely, thank you very much. However, the Directory app apparently still takes it's list of extensions from the voicemail.conf table. That's not so nice, since it means maintaining the same list in two places. Am I missing something, or is there a version of the Directory app that queries the users table instead of the

For some time (since pre 1.0), I've been seeing the following messages fairly regularly from some, but not all, of my SIP devices: Nov 23 06:37:59 NOTICE[2568]: chan_sip.c:7645 handle_request: Registration from 'John Doe <sip:9005552368@100.200.200.100>' failed for '200.100.50.25' I have a mix of Sipuras, Grandstreams, ZIPs and Ciscos, but the message seems to come