search for: oneunified

What types of ISDN BRI cards work well in Europe (Guadeloupe, Martinique and France) ? For example: AVM C2 or AVM C4 or eicon Diva server 4 BRI? Any others? Which driver is appropriate? Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2003111...

i'm looking at potential asterisk service providers and came across oneunifed.net i googled for opinions and feedback, but haven't come across anything yet. is anyone using them or does anyone have feedback on their asterisk support and expertise? tia, george

...Would you recommend it for a production server? > > Obviously, if this works, this makes for a cost effective platform where > you obtain one E1/T1 to a provider, and they can provide TDM and data over > the one circuit. No separate router required. > > Ray Burkholder > ray@oneunified.net > http://www.oneunified.net > 704 576 5101 > -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/atta...

Anyone know a sip-client that will work on a Handheld PC running WINCE for HPC. I can find some for PocketPC, but the wont work on my HPC ?? /HHA _________________________________________________________________ Scope out the new MSN Plus Internet Software — optimizes dial-up to the max! http://join.msn.com/?pgmarket=en-us&page=byoa/plus&ST=1

...do I do that? I already have a tftp server for my SIP based phones. Do I need a tftp server for skinny configs at all? And if so, can it be the same tftp server as the SIP ones use (I'm not sure about this as the OS79XX.TXT makes newly arriving phones load the SIP image). Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2003110...

...that requires an open connection to the backend DB. Flash is very nice as it loads quickly and theres no applet bogging things down :) We are currently using this flash technology for various other things. It has lots of applications. S "Message: 1 From: "Ray Burkholder" <ray@oneunified.net> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] Anyone, ideas for incoming call management for CRM system Date: Wed, 31 Dec 2003 11:53:49 -0500 Organization: One Unified Reply-To: asterisk-users@lists.digium.com You may wish to try Java. It is a client OS indepe...

Hi, Does anyone know of Canadian VoIP termination providers? I have Canadian customers and would like to provide Canadian dial in and dial out (canadian callerid). Thanks!

...on belonging to it via the host= line in the sip.conf file without user authentication, or C) some other way I have yet to fathom I'm trying to differentiate between legitimate gateways that initiate calls vs other gateways that should get a very limited inbound capability. Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 644 6999 x2002 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean.

I assume there are several people on this list that have Cisco Call Manager implementations under their belt.... We are beginning a call manager implementation and the first question I asked Cisco was, should we use SIP or Skinny. Cisco is pushing me towards Skinny, saying that I will lose some functionality with SIP. They also say that most of their customers implement skinny. I see two

Jeremy, In some posting in the mailing lists, you mentioned that docs for h323 had been submitted but hadn't made it into distribution. Could you post those docs in your download directory? I'm trying to understand the nuances of your driver, gnugk, and extensions. Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2003122...

...): Keyword h323 does not make sense in type=h323 Why is that? 3) When making a call from an h323 client such as ohPhone registered with gnugk, I can make the call, but it uses the context from the [general] section, rather than the context in [office]. Is this supposed to be? Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean.

Hi All Will cisco 7910 ip phone compatible with Asterisk? I know that 7960 are fine. David Kwok -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e8023f35/smime.bin

Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/aa4eda3c/attachment.htm

I need to get 2 Asterisk servers working together. I have been reading and doing just about every example I have been able to find here on the list and the Wiki. It's now gotten to the point that nothing on box2 seems to be working. I seem to have a major problem understanding the format. Here is what I have so far. It's 3 days of hair pulling and nothing seems to work! Asterisk box 1

...capability (user can take own extension [logically or physically] between offices) b) local * server at each office to handle trunking to hosted * c) central auto attendant for all offices D) local extension to extension calls stay local (don't cross the WAN to the hosted *) Ray Burkholder ray@oneunified.net http://www.oneunified.net 704 576 5101 -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean.

...oot scheme: export LANG=C or at least execute it before running perl scripts. Redhat has EVERYTHING set to LANG=UTF-8 and it screws up all sorts of perl stuff, and several other pre-written programs in other languages too MATT--- -----Original Message----- From: Ray Burkholder [mailto:ray@oneunified.net] Sent: Friday, October 17, 2003 5:16 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] AGI problem (crash) in RH9 You may wish to upgrade your kernel to 2.4.20-20.9 through 'up2date'. Regards, Ray Burkholder > -----Original Message----- > From: asterisk-u...

...a part number SW-SM-UL-7960 for licensing SIP on their CP-7960 phones. Is this actually required to be purchased to keep everything on the above-board when using Cisco's SIP phone with Asterisk, or is this for something else? Ray. -- Scanned for viruses and dangerous content at http://www.oneunified.net and is believed to be clean.

Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These

Hi, I'm trying to detect and or log the duration a a conference (Meetme). I need it in order to do some billing for theses services. Any ideas on how to do it? I googled around but found nothing. Thanks in advance epablo -- Pablo Endres <epablo@comvoz.com> ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199

I'm not really looking for working configurations as much as I am looking for people who can say "This is a solid product and I trust my business to a solution running Asterisk." As far as pre-sales work... Well, tell that to my consultant. I'm quite excited about *. I've got my company sold on it, they just want some reassurance that it's ready for prime-time