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Hello, I''ve published a Q&A in the Vancouver.rb series with Brock Whitten. Questions include: * How did you get started with Ruby on Rails? What makes Ruby on Rails a great choice for developing web applications/sites? * Can you tell us some challenges you''ve faced developing using Ruby on Rails? * Off the Rails - Can you tell us a little bit abou...

Hello, I''ve published a Q&A in the Vancouver.rb series with Brock Whitten. Questions include: * How did you get started with Ruby on Rails? What makes Ruby on Rails a great choice for developing web applications/sites? * Can you tell us some challenges you''ve faced developing using Ruby on Rails? * Off the Rails - Can you tell us a little bit about...

Has anyone ever gotten * to work on commercial unixes such as HP-UX, Solaris, AIX? What about other architectures than x86? -- . -----BEGIN GEEK CODE BLOCK----- Version: 3.1 GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h---- r+++ y++++ ------END GEEK CODE BLOCK------ . -------------- next part

...even though im not using it now. also, what about the acd functionality of asterisk? we have two acd groups now with at most 5 members in each logged in at a given time. call distribution is round robin and cisco does have some nice call reporting features with the icd software. -- Chad Whitten Network/Systems Administrator cwhitten@nexband.com 601-944-4801 Phone

are there any open source sip softphone (Window OS version )?

...002,2,Dial(SCCP/Test2,10,tr) exten => 1002,3,Voicemail,u1002 exten => 1002,102,Voicemail,b1002 exten => 1003,1,SetCalledParty("Test3"<1003>) exten => 1003,2,Dial(SCCP/Test3,10,tr) exten => 1003,3,Voicemail,u1003 exten => 1003,102,Voicemail,b1003 Chad Whitten Network Administrator neXband Communications cwhitten@nexband.com 601-944-4801 Phone 601-944-4803 Fax

Hello everybody, Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc. Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.

I have Asterisk setup in a SOHO environment. I have 2 X100P cards at Zap/1 and Zap/2. I have 1 TDM400P card with Zap/3 - Zap/5. I have subscribed to callwaiting, callerid and calleridcallwaiting from Qwest on the 2 PSTN lines - Zap/1 and Zap/2. My problem is when I'm in an active call to the outside thru Zap/1 or Zap/2, I can't pickup the incoming callwaiting call. I can see the

...tation that would indicate if Asterisk supports H.248 and everything I have come across seems to indicate I should use MGCP which I would agree is a better choice but unfortunately the equipment I am trying to integrate only does H.248. Could anyone point me to something related to this. -- Chad Whitten Metro Network Solutions (601) 366-6630 Phone (601) 366-6066 Fax (601) 842-6804 Cellular (601) 519-4172 Pager cwhitten at metronetsys.com

Hi Guys, I have three Samba file servers running different versions of CentOS and Samba. They are all joined to a proper Windows 2003 domain. I'd like to migrate the WIndows domain to 2016 and all the Linux servers to CentOS 7. I can do Windows 2003->2012->2016 without too much trouble. I'm concerned about the Samba side. The servers are all using extended file system

Got at suggestion from CarrierAccess to use the Adit 600 as an VoIP router using MGCP IP protocol, instead of controlling it through an E1. Have anyone tried this configuration? How does MGCP works? I've tried to search for it on Google, but I only find the protocol specification for it. Is Asterisk fully capable of this? I can't find any documentatin covering the use of MGCP in Asterisk.

Hi Scott, Am 12.02.2018 um 06:46 schrieb Scott Whitten via samba: > I can do Windows 2003->2012->2016 without too much trouble. Can't you join 2016 to the 2003 domain, demote 2003, and raise the functional levels? Or join a Samba AD DC and continue without MS. ;-) > I'm concerned about the Samba side. The servers are all using...

I have an audiocodes mp-124C/FXS gateway that was a leftover from a failed voip experiment and I have no documentation at all for it. Anyone have a manual in pdf form they could share? -- Chad Whitten Network Administrator neXband Communications cwhitten@nexband.com 601-944-4801 Phone 601-944-4803 Fax

...the two. sip show registry Host Username Refresh State metaswitch.nexband.com:5060 voicemail.ne 120 Request Sent i tried voicemail@metaswitch.nexband.com & voicemail.nexband.com@metaswitch.nexband.com with the same results any suggestions? -- Chad Whitten Network Administrator neXband Communications cwhitten@nexband.com 601-944-4801 Phone 601-944-4803 Fax

Has anyone had any success using the Adit 600 with the CMG card talking MGCP to asterisk? I want to have a central asterisk server with 10 Adit 600's at various locations providing 24 FXS ports.... Thanks, Isaac

Anyone seen or tried this yet? http://www.voip-weblog.com/50226711/uplink_connects_sip_skype.php Michael

What is the deal with "CSR's"? TWTelecom is telling me that I can't port a number to their service without a Customer Service Record. Apparently this is easy with Verizon, and not so easy with some other companies. Basically I'm at a brick wall with a couple of ports because TWTelecom is telling me I HAVE to get a CSR and certain other providers (Time Warner Cable for

...multiplexed terminal node controller (TNC) This was the very thought behind "trunked" communications around 800mhz. Gee ... there are a bunch of cell phone frequencies just above that and 3x800 = 2.4g and .... I'll leave the possibilities to the imaginative reader On 9/9/05, Derek Whitten <derek@kfuq.net> wrote: -----Forwarded Message----- From: Mike M <no-linux-support@earthlink.net> To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM Date: Fri, 09 Sep 2005 10:26:09 -0400...

A customer of mine wants an IVR where the first 3 choices are 1 English 2 Spanish 3 French I can build the IVR but how do I get the system prompts to then speak the selected langauge. For example, a caller has selected Spanish and so is routed to the Spanish part of the IVR. At some point he breaks out of the IVR to leave a VM. How does the system know to continue offering him Spanish?

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'