search for: raynettech

...uly is SIP based, and if so, managed to get it to interoperate with Asterisk. If so, it would prove to be an interesting paging mechanism and I would think would have immense value to any organization with multiple mobile individuals. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226

...nnel.c: Avoided initial deadlock for '0x9896848', 10 retries! attempting to stop asterisk from the CLI causes the CLI to become unresponsive and a trace shows chan_sip goes into a mutex_wait state. Anybody seen this? Have a fix? Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070110/8b9e87f4/attachment-0001.htm

...ase. Can anyone explain to me exactly why this is. I don't really mind buying more licenses if I need to but I can't seem to wrap my head around where the Codec translation that is requiring the license is taking place. Regards, Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060302/8658fee2/attachment.htm

Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten => 9,1,Dial(Zap/g1/<CELLPHONENUMBER> where <CELLPHONENUMBER> is the number it is calling out to. When option 9 is

...;; reload) action "Reloading ztcfg: " /sbin/ztcfg ;; *) echo "Usage: zaptel {start|stop|restart|reload}" exit 1 esac exit $RETVAL Hopes this helps anybody else trying to implement on a FC3 base. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 693-2226 x 31 Toll Free (877) 693-2226

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

...be a bit of buzz on the list about ADSI phones and their configuration, but no clear progression of what really needs to exist to have a basic config. Could someone please post what they had to do to get an unlocked ADSI phone to work? Thanks Raymond McKay President RAYNET Technologies rmckay at raynettech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20030305/792694bc/attachment.html>

...If anyone can step up on this one and provide information, I would be happy to put it all together in a fully documented HOWTO so that anyone else attempting this configuration will have a clear and concise guide. Regards, Raymond McKay President RAYNET Technologies LLC (860) 833-9720 http://www.raynettech.com

Are their any issues with Asterisk and Redhat Enterprise? I have see one or two posts with issues concerning compiling zaptel drivers but that is about it. Just looking for some consensus to if any problems exist with it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040422/56d03b72/attachment.htm

does anonybody know what is the status of gsm codec in snom phones ? they were some issuses in archives, some problems so i would like to know what is the actual status. best regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ majo at sunteq dot sk

Hi folks! Based upon your experience on the field what wifi sip phone would you reccomend ? A customer asked for a wireless * install and I'm looking for advice, tnx Alessio Focardi [[*] - Interconnessioni Italy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060628/05b2fb30/attachment.htm

Hi, How difficult is to setup and maintain an Asterisk PBX with phones from multiple vendors? Is it even worth considering or is it safer to pick one vendor for phones and stick with them? I am more concerned about proprietary DHCP extensions, firmware upgrades etc..If anyone has any thoughts or experiences they would like to share I would be more than happy to hear from them. Thanks -Ravi

I have been struggling with this issue for about a year now. There were just too many IP phones to choose from at all sorts of price points and not enough information about any of them. Now I am looking at the situation again and if anything it has gotten worse. There are even more phones and all sorts of opinions. For every person that says phone x is great there is someone else complaining

I'm hoping that someone else has seen this problem. Running CVS version from 03/08/03 20:00 T100P -> Zplex-10 All internal call routing seems to work fine. Atempts to make calls on the FXO interface fail. Asterisk picks up the correct number dialed but there seems to be a problem with the Zplex transmitting that number on the POTS line. All attempts to dial out fail with the LEC giving

Hello, I've created a patch to correct error responses for the MeetMeList manager action. Currently MeetMeList produces an error if no conferences are active, success if any conferences are open. Requesting a conference that is not active while other conferences are active does not produce an error. https://issues.asterisk.org/view.php?id=18141 With the patch