similar to: Follow-me to my answering machine :-(

Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio -> audio in to jack, audio out from jack -> current Asterisk application Outgoing call audio <- current Asterisk application However, I need

Hi, Thanks for your info, What is the impact of the following line in dialplan, Dial(SIP/19201/19202,300) On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03

Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and "This user is temporarily unavailable". Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as

Hi, I have been headbanging with asterisk and Jack for a while, decited to ask other linuxists for an advice. The problem is that Jack is compiled from source (0.118) in /usr/local/, but menuselect says "XXX" for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide

Hi, List, I am looking for a cheapest (and therefore most funny) way to attach GSM card to my asterisk home box. Needed features: Calls+SMS in/out one or two SIM cards (ports) Should I try looking for a GSM PCI card that is compatible with linux/asterisk, or GSM USB card, or modern full-blown SIP GSM gateway (with ethernet)? Maybe an ordinary cell phone with USB interface and mangling with

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi List, Our company has several small distributed offices we would like to inter-connect with bridged VPN a single subnet (last example in http://www.shorewall.net/OPENVPN.html). We have SIP phones in every office (up to 5) so we can use SIP without any NATing and securely. Max theoretical simultaneous calls possible ~30, but we have ~5-10 @ regular basis. OpenVPN server would be in the same

Hi all, Anyone know why this happens? Mem: 524288k total, 508120k used, 16168k free, 0k buffers Swap: 0k total, 0k used, 0k free, 0k cached PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1 root 15 0 2152 664 576 S 0.0 0.1 0:49.26 init 7398 root 18 0 10172 2904 2312 S 0.0 0.6 0:00.21 sshd 9856

Hello there, I have successfully installed and configured asterisk for use as an office PBX using SIP trucks and Voip handsets (using g.729 codec) which works great. Now I wish to try and configure asterisk to do a HTTP request and submit callerID to an external website when a call is missed. eg Someone calls PBX and rings extension 100 -> Call is not answered -> HTTP request is initiated

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

In followme , is it be possible to have a third option.... Whereas, takecall=>1 declinecall=>2 proposed option transfercall=>3 or, transferring the call directly from followme isn't really neccessary, if the callee could answer the call, and transfer it someplace, that would work as well.... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Folks, I want to setup a follow me routine so that asterisk can call me on the multiple numbers. I tried some of the samples at voip-info but there is a problem with those examples. I dont have coverage in my home area and my cell phone answering machine picks up the phone right away so my home phone never rings. I also want the caller to be able to leave a voicemail and the cell phone

The Asterisk Development Team would like to announce the release of Asterisk 18.19.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.19.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Asterisk 18.19.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.19.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Asterisk 20.4.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.4.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of Asterisk 20.4.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.4.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

Hi, Is there a way to check on AMI if a user is currently engage on the phone? i would like to display on my portal whether a user is calling or not. thank you regards Ron

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

Hello, I saw that Asterisk don't calcultate fine the ANSWEREDTIME. I want that when ANSWEREDTIME =~ 5.6 become 6 and if =~10.3 become 10 because, now, if ANSEREDTIME =~ 15.9, it become 15! it isn't correct How can I have a rounded ANSWEREDTIME ? Where have I to manipulate the sources? thank you -- Francois -------------- next part -------------- A non-text attachment was scrubbed...