similar to: Early media recognition

hi, I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well. At this moment I

hello, I'm trying to use a remote SMB/CIFS share to backup a local directory tree. I use rsync "version 3.0.4 protocol version 30" and mount.cifs "1.10" with a kernel 2.6.18. The directory I'm trying to backup is quite big and has a lot of files inside. [root@linux dir]# du -hs . 50G . [root@linux dir]# find . -type f | wc -l 41929 I get the following error when

On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > I added the bind_rtp_to_media_address=yes on all endpoints but still the > same behaviour. The funny thing is that the G711 audio early media works > and doesn't have that Private IP issue. I was also able to cross check with > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following > capture (PJSIP): As

Hello, I have the following scenario: 1. VoIP Gateway G400 connected to PSTN 2. Asterisk server 1 (working as IVR) 3. Asterisk server 2 (working as ACD, with several agents connected) I have incoming calls coming from PSTN through the VoIP Gateway to Asterisk server 1 (IVR). When the IVR ends working with the call, transfers it to the Asterisk server 2 (ACD). In Asterisk server 1

Hello, I'm trying to find more information about this Remote Attended Transfers, as is explained in https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers for Asterisk 12 using pjsip stack Was Remote Attended Transfer implemented in previous versions of Asterisk (versions without PJSIP, Asterisk 11 and previous)? Where can I find configuration examples to do it work

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

Zed / List, On my workstation I''m running lighttpd 1.4.11 and mongrel 0.3.12.4. Here are my proxy settings: proxy.debug = 0 proxy.balance = "fair" proxy.server = ( "/RailsAppSample" => (( "host" => "127.0.0.1", "port" => 1337 ))) I can access http://localhost/ (default lighty welcome page) and http://localhost:1337/

All, I realize this was probably off-topic (not really mongrel-centric) -- my apologies. I "solved" the problem by appending to my Environment.rb: ActionController::AbstractRequest.relative_url_root = "/RailsAppSample" How come that doesn''t serve up the default "public/index.html" ? It caused a Routing Error for "", so I added a temporary

I''m a noob to rails and i could use some help. Fedora Core 5 Ruby 1.8.4 Rails 1.1.2 Postgres for My Database Using webrick for testing at the monent but have lighttpd installed but the service is diabled. Here are my steps that i fillow. In my home directory i type rails test #cd test #sudo chmod -R 775 log #sudo chmod -R 775 public #ruby script/server Then i make my postgres database

Hiall, I have two modules for my controllers containing the following files /app/controllers/admin/new_controller.rb /app/controllers/admin/list_controller.rb /app/controllers/admin/sidebar_controller /app/controllers/community/input_controller.rb /app/controllers/community/sidebar_controller.rb Each of those controllers is declared with the appropriate module like so class

How does one go about troubleshooting a "Routing Error" "Recognition failure"? I''ve used the Webrick log to get down to a "recognize_path(path)" call in /actionpack-1.12.1/lib/action_controller/routing.rb and put a logger.debug path there, but that just told me what I already knew, and the definition of "recognize_path" looks like its

How can I handle ''recognition failed'' errors so they act like typical 404 errors? By displaying a message, page and/or redirecting. thanks csn __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hi all, someone has used the voice recognition software named Sphinx??? Can he tell me how to use and its performance??? Thanks Marco -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090408/4acd09c5/attachment.htm

Today I have a problem with soundcard recognition. In winecfg, the alsa-driver MIDI Out and In devices are: Midi Through Port-0 ?? I have done everything I have done in the past to get sound in wine. I have purged pulseaudio. asoundconf list give me: Generic and I have set that to be my default card. How come wine is not recognizing it? Susan

Hello! I'm interested in getting a Chinese optical character recognition program to run under wine. http://www.twinbridge.com/Products/SharpEye/ocr.html According to the web site, it may run on Windows 95 as well as the old Windows 3. Is it therefore safe to assume that it will run under wine? Surely it should be able to handle anything that can run on Windows 3 right? Any advice will

Hello list, I'm wondering if anyone has had success running dragon /scansoft/ naturally speaking voice recognition software with WINE recently? Thanks! Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.winehq.org/pipermail/wine-users/attachments/20030609/86e6dd84/attachment.htm

Hi all, We have a project involving voice recognition, and will need a vocabulary of 10,000 words (actually names). Can anyone recommend a product that works with Asterisk? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed...

I have been having problems dialing into Verizon conferencing using my * system. If i dial using a POTS line directly, the dtmf codes for the conference room are recognized with no issues, however, verizon doesnt recognize the keys when i press them as being a valid code. I remember something a while back (unfortunately - lost all my IMAP email) - Something about verizon lines and dtmf and

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

Hi, I connected my Wildcard X100P to the PSTN and created a context in extensions.conf which rings a number of SIP phones on inbound calls from the PSTN. When I compare the actual PSTN rings with Asterisk recognition of the incoming call, Asterisk rings my SIP phones on the third ring of the incoming call. Here is some log info: On the first ring: -- Starting simple switch on