search for: brucek

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

...I am running CVS-HEAD-05/06/04-17:52:43. TIA sip.conf: [7752365815] type=friend username=xxxx5815 secret=xxxx host=dynamic mailbox=7752365815 context=vpbx-wpti qualify=1000 dtmfmode=inband disallow=all allow=ulaw allow=alaw nat=yes voicemail.conf: [vpbx-wpti] 7752365815 => 0000,Bruce Komito,brucek@wpti.net Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:...

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

...nces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Darren Nickerson Sent: 04 October 2004 16:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] echo cancellation: the never-ending quest fortruth "Bruce Komito" <brucek@bagel.com> spake thusly: > 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...

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

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)

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,

...asterisk-users-bounces at lists.digium.com] On Behalf Of Bill = Michaelson Sent: Friday, October 24, 2008 13:41 To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Sonicwall potentially causing long ping = timesto SIP phones Kristian Kielhofner wrote: On 10/23/08, Bruce Komito <brucek at bagel.com> wrote: =20 > We've had LOTS of problems with Sonicwalls doing bad things to SIP and = RTP > connections. I've seen the delay thing, as well as the Sonicwall = throwing > away entries from the ARP table because of inactivity. I've also = seen > sporadic...

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:

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.

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

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 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

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

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

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

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?

Hi list, I am planning a migration to virtual machines, and was considering with it to move from 1.4 to one of the later versions. My and my clients' 1.4 setups have been rock solid and I don't want to put myself into any unnecessary trouble. Those of you with solid experience with all these versions, what would you suggest? What new and exciting enhancements would newer versions bring

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