search for: komito

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

...ent 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) 236-5815

Is any one else having problems with Voicepulse today? Suddenly, I can't register and calls to my Voicepulse numbers get a fast busy. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

...ntly 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

...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. www.servers-r-us.com (775) 236-5815

...* waits for call progress from the gateway before giving the caller the appropriate indication, i.e., ring or busy tone? I have been told this is a result of setting * to "forced ring" and this should be turned off, but of course, on * it is probably called something else. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

...problems coincidental with the messages. I am experiencing a large number of dropped calls, but the dropped calls occur both on PRI and non-PRI calls. Has anyone observed these messages, and if so, do you know what they mean and are they something to be concerned about? Thanks in advance. Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

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

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

...ng to the * log, the sound file is being played, but no sound and no errors. Has anyone ever seen a problem like this? It's almost as though I'm missing some software, but I can't imagine what that would be there are no errors reported, either at install or run-time. TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115

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 sip device (Supura) does n...

...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'm doing something wrong. Anyone have any ideas? Thanks in advance! Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115

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

...he 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 else seen a problem like this, and if so, what's the solution? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

The most recent version of GSConfigure is available at www.buffalo.edu/~sbesch Several serious bugs that kept the program from getting started have been ferreted out and corrected with the help of Bruce Komito. The program is now actually running on someone's machine other than mine. I have built this version with the oldest copies of the system dll's that I could find inn an effort to solve the VB setup bug, so, hopefully it will no longer send anyone through multiple restarts. You should ha...

...t I don't know where to start. I have nat keep-alive turned on, but I had to turn stun off because it was causing a long, inexplicable delay after dialing before the call would complete. I'm realizing NAT with VoIP is a real problem. Anyone have a silver bullet they wish to share? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115

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

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

...ser 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 that anyone has ever done this. If you have, please respond. Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115

...That's not so bad with the auto-attendant, because it answers right away, but it's kind of a problem for the other DID numbers. I stuck "Ringing" statements every I could think of, but to no avail. Is there something specific I must do to cause the caller to hear ringing? Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 284-5800 ext 115