similar to: Connecting 1-2 GSM ports to asterisk?

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

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 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

Is there a database of MAC address prefixes used the common VoIP devices. I see the Linksys Sipura devices state with 00:0E. Does the same apply to other Linksys VoIP equipment? Is there some way VoIP equipment allow themselves to be identified by requesting data from some ports?

?Please <I would like to get informatin regarding the stream line for an interenet radio, Thank You Sandy Flores ?(714)963-7462 Bus (714)438-0156 Fax -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/icecast/attachments/20130127/1a46082c/attachment.htm

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

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

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

Hi there, I just installed the GUI for Asterisk 1.4.4 and correctly set my settings but when I use my browser to access it, it gives me an error saying "Not Found. Nothing to see here, move along" with "asterisk" in the header and footer... anyone had this problemn before? greetz

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

On our CentOS 7 servers I see there is a new kernel available when doing yum update: kernel 3.10.0-1062.12.1.el7 kernel-devel 3.10.0-1062.12.1.el7 However I am not able to find any announcement at redhat.com for this kernel. The only announcement I find is this from CentOS: https://www.mail-archive.com/centos-announce at centos.org/msg11573.html However that CentOS announcement links to

Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that

To R Support Team, ONS would like a restricted number of its IT staff to view the source code for the latest version of your software, to check it against our source code security guidelines.The source code will be securely stored with access limited. ONS is quite happy to agree that we would not - copy or change your source code without your agreement - share the source code with anyone

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Executing [123456 at NPDB2:76]

Hello, Asterisk party, If block the call before dialing (Hangup()), CDR's don't write to MySQL or CSV. Otherwise, if Hangup() is after Dial(), CDRs write normally. Here is the dialplan: ; we skipped dial, because the number is "blocked" exten => _X.,n(Finish),Hangup() exten => h,1,NoOP("hangup") exten => h,2,ResetCDR(w) exten => h,n,NoCDR() exten =>

I''ve got a linux machine (fedora core 3) with 4 network cards. I looked at the howto and the only example that is close to what I need to do is section 4.2 on multiple uplink providers. I feel like I''m so close but just can''t get my head around the final part. Here is what I have eth2 and eth4 connect to 2 different isps. I want all connections the come from my dmz

On Aug 7, 2007 'Mojo' wrote: Nicholas Blasgen wrote: > I've got 4 SIP phone lines with a call-limit of 2 for each. I've > written a handy macro to allow my users to dial a phone number and the > macro will figure out the next available line to use by first checking > if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a > backup, and if it

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

Hi, I have an xlite registered with asterisk server. When i dial a number AGI is invoked. and in this we are running to threads one to record files and one to play files. So i dialed the extension and i started recording and playing at the same time. On the xlite i m getting an indication when recording my voice and at the same time i could see playing the other file too. But in the directory