similar to: CDRs, Conferencing, and MeetMe

I am trying to use festival (latest version 1.4.3) I have downloaded all the files needed and patched it with the provided diff. festival does work and does tts fine. but when I call Festival either from an extention or an AGI script, I get this in my asterisk messages log, but no sound on the channels (H323 or SIP) - they (the clients) just say "trying" and then hangup... Sep 24

I am just now investigating Asterisk. Can Asterisk provide 6-10 party teleconferencing when configured properly?

I''ve got a brand Polycom Viewstation FX video teleconferencing unit. I''ve got a Shorewall firewall 1.4.9 instance running on box with a 2.4.20 kernel. I can not receive H.323 video. From the FAQ I''ve read that there is an unsupported H.323 connection tracking module that is no longer maintained. What options are there to make H.323 work with a IPTables based

Just installed the new 4-port FXO card and moved two pstn lines from existing x100p cards to ports on the FXO card. All zapata.conf entries that were functional on the x100p's were copied to the new FXO channels (including callprogress=no). Observations thus far: 1. asterisk will spontanously decide a pstn call has arrived, and ring the sip phone designated in the dailplan. Verified

Hi all. I have a strange problem, I've got a AS5350 hooked up to a telco using two trunked E1's The 5350 should only act as a GW to a sipproxyserver. THe thing is it seems to be only oneway audio? There are no firewall at all, and the audio still only get one-way When I call from pstn --> as5350 --> sip-sip-phone I can here the sip-phone ,, but the sipphone cannot her the

On your web you have a link http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard To Setup Calling with Diamondcard.us and I signed up and paid the money according to Stephen Karrington it was all automated... And it was automated to take money but when you look for service hookups or information you don't get it. I have tried now for last little while to contact them for support

Just came by this link So I'm posting to keep the community informed. I don't use or endorse this product. I'm just letting people know about it. http://www.indosoft.ca/audioconferencesystem.htm Audio Conferencing System & Teleconferencing Solution that connect seamlessly over TDM and IP networks. This audio conference system include a comprehensive set of features and easily

Is this supported? I have a very simple setup where I have 2 X100P cards and a TDM10B. The TDM10B is connected to a phone that has a digital answering machine built into it. If I make an inbound call on either X100P interface it gets transferred to the TDM10B interface. If I let it ring the TDM10B interface answers the call and the greeting message of the answering machine starts. Then shortly

I''m trying to store an xsl stylesheet in the database and return it to the client, but at some point in the process all the angle brackets, etc are parsed out of the xml, so I get &lt;defaults&gt; instead of <defaults>. Anyone have any pointers how I would go about turning off that behavior? -Derek

Hello, I am trying to read in an Excel file that I saved as a .csv so I can analyze my dissertation data! I am getting really frustrated because this is what I keep getting: In read.table(file = file, header = header, sep = sep, quote = quote, : incomplete final line found by readTableHeader on 'month.csv' can anyone offer some help? Thanks a lot! catherine [[alternative HTML

hi, im looking at deploying asterisk in a small corporate enviroment which will have approx. 1200 IP Phones and an average of about 100 to 200 calls at any given time. The calls will be sent out SIP to my Cisco Gateway. Im running Asterisk on a Dell Dual P3 1.2ghz running Fedora. Is there a calculator or a spreadsheet which could tell me about how many calls I will be able to make through *

Hi, Is there some script which can be called from a * extension to playback the recent incoming callers on a particular PSTN line? In the UK 1471 is a BT number which plays back the most recent callers number, it also gives you the option to call this number back (now charging you for this service too!). Is there anything similar in asterisk-land? thanks Mike

I just received two TDM400P cards, but I'm having problems with them. The full info is at: http://pastebin.com/824079 Extra, I'm using : libpri-1.2.3 zaptel-1.2.10 On a x86 stable Gentoo box. Kernel: 2.6.17 gcc-4.1.1, glibc-2.4-r4 Is that an hardware problem? Should I try the other card? -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog:

Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist

hi, ive read through the last few posts with people having problems compiling the asterisk-addons for mysql support, and none of them have helped me resolve my compile problem. I currently have -- CVS-06/24/04-22:20:31 and downloaded asterisk-addons. I compiled * first then asterisk-addons, have added CFLAGS+=-I../asterisk/include When I try to make install for asterisk-addons i get

I have managed to get spandsp working, and if I dial a specific extension I can receive faxes. WhooHoo. However, I was wanting to use the "fax detect" option in order to allow individuals to receive faxes, but can't get that to work. Given the following extensions (mainly copied from examples on the wiki), why is the call simply passed onto the sip device rather than being

I followed the Wiki instructions to get zaptel to work on Fedora core 2. It looked like everything went perfect including the loading of ztdummy. However, I am having meetme and MOH problems synonymous with ztdummy not loading. Take a look at my lsmod...Any ideas? (I am running stable Asterisk on a DL360 - Dual processor) Module Size Used by snd_pcm_oss 46201 0

Is there a way to get asterisk to send you a email when it looses or an extension doesn?t re-register Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 Website: http://www.upperclassman.net Billing Questions: billing at upperclassman.net Rental Questions: rentals at upperclassman.net Maintenance: help at

On Fri, Jul 17, 2015 at 9:17 PM, Wayne Workman <wayne.workman2012 at gmail.com> wrote: > I must say that I've learned an incredible amount and only hope to learn > more. Keep up the good work. Thanks. > I have been prowling your email correspondence for the FOG Project. You may > find them here: https://fogproject.org/ and their forums are located here: >

I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options