similar to: determining cause of dropped calls?

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

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)

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

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'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 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

Has anyone tried out the Linksys RT31P2 with Asterisk? Seems like a really great solution for remote users... even supports QoS. Too bad it doesn't also have VPN functionality built in. Here's a link to the product: http://www.linksys.com/products/product.asp?prid=652&scid=29 -Ron -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3234 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090226/a46e68fa/attachment.bin

Hello, I was running a very simple test setup with * HEAD 7/15/2004 on Fedora Core 2 and things were working fine. Today I upgraded to RC1 and my asterisk service will no longer start. I downloaded the tarball, extracted, ran 'make', ran 'service asterisk stop', ran 'make install', removed all files in /etc/asterisk, ran 'make samples' and then ran 'service

I have just installed * (RH9, P4 3.0GHZ, 1G RAM) in a small office, using three TDM400's with 4 FXO's each for incoming calls. Outgoing calls are (for the moment) routed via VoicePulse. Phone sets are Cisco 7940G's using SIP. I'm getting intermittent echo on outgoing calls, and my understanding, based on reviewing the wiki and several posts here, is this: >>>> The

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

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

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

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 am investigating the deployment of VoIP/* in Eastern European areas where there is no PSTN infrastructure. As you can understand DSL/Cable connections are a dream. The only option is satellite. Does anyone know of any satellite providers that have low enough/acceptable delays for VoIP? Thanks, Yiannis.