similar to: DTMF forwarding and Page

Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone

I'm building a paging system composed of roughly 10 switches in daisy chain, with an embedded box with a speaker and a microphone for each switch. The embedded box runs my software. I need the system to be resilient to any network partition, so that anyone can send announces from any mic to all the reachable clients. I'd need also to page a subset of all the speakers. I'm

hi all, please help... I am calling in the simplest way among two linphone clients connected to one asterisk server... the call ends on one side without any sign of problem, while on the other side it stays connected. I checked the SIP dialogue and at some point the server sends a BYE message to one party I have no timeout set, though the duration of a call is always around 20s. the two

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

I'm trying to improve Opus on an AMD Geode CPU, which has limited SSE support (called 3DNow!), but MMX. Without optimizations I can only encode 16 bit audio @16KHz with complexity up to 2-3 without underruns. I tried compiling with SSE2/4 optimizations, but all I got was a crash with SIGILL, so I looked into optimized code and found that a good starting point was the dot product, so I

Dear list, I have been using asterisk for some time now. However I have never used it with any of the digium or compatable cards (Purely used for SIP). I understand that for using Meetme, I need to have a timing device, which could either be hardware or zrdummy etc (I am not using any right now). Can someone tell me if the timing device is needed for voicemail and other applications too?. I am

I had unexpected hangups from my asterix box using the i4l driver. (SIP <-> SIP calls worked execellent, but SIP<->ISDN didn't.) Then I changed the i4l driver in modem.conf with the chan_capi from jungham. (http://www.junghanns.net/asterisk) I followed his instructions in the INSTALL file, and first encountered some errors compiling it. It help by deinstalling several

Hi, Is it possible to create a meet me room on the go through dial plan? I am looking to use AMI Originate to drop a call into meetme room and once it's proved that party is joined, play him an announcement, grab few numbers from them, and then dial a second number and drop into the same meetme room. The reason to use this is to be able to know when the channels connected because both

On late august, there was a thread about setting up some meetme conferences to be able to follow Astricon remotely. This indeed could be nice for those that can't attend for various reason. And of course is a demonstration of Asterisk capabilities... :) (Astricon without a remote conference for guest is like a big it expo without internet connections...) I have some bandwidth here, so can

Is it possible to play and audio file into a meetme conference for both parties to hear? I thought I remembered reading something about it, but I can't find it now. Any help would be greatly appreciated. Paul

I've set up asterisk with my X100P as a home answering machine. Works great so far - answers the phone after 20 seconds, runs the phone tree, emails voicemail, etc. However, the one feature traditional answering machines have that I haven't been able to figure out is how to listen in on the call. Ideally I could just route through Console/dsp and hear it on my speakers. I've tried

Hi All, The meetme.conf have created as below: [rooms] conf => 101 conf => 102 and extensions.conf as below: exten => _1XX,1,MeetMe,${EXTEN} why the warning printed when I called 101. WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application 'MeetMe' for extension (ipcentrex, 101, 1) And I found asterisk have not load the meetme.conf when it starts up.

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

Hi, folks. I have a problem using Asterisk 1.2. I create conferences using app_meetme and Zap channels, and for every participant I run the script defined by AGI_BACKGROUND_SCRIPT to be able to listen and react to DTMF tones. As the docs tell me, when using the AGI background script one loses the ability to control the meetme conference via the command line so for muting conference participants I

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

Good day all, we have following setup: Debian Etch 64, Asterisk SVN-branch-1.4-r66244 with mISDN 1.1.3 and 2 Digium cards B410P. Our customers calls in the office through ISDN lines and then get a possibility to join meetme conference. It works well except when customers are using SIEMENS phones (not IP): DTMF is not recognized. Does someone have an idea on what could be the problem with

Hi, First of all I am not sure that what I am trying to do is correct/supported, but here is what I'm trying to test: Some of my endpoints only have g723 codecs. Because of this I am only allowing g723.1 codec in sip.conf and h323.conf. Calls between endpoints work fine. I am trying to configure voicemail and meetme applications. I see that all voice files in asterisk are in gsm format and

In this message I try to summarize what I have learned in these last two weeks. My primary sources of informations were the * list archives and linux ISDN docs. I ain't no * master, so don't trust too hard. Relevant messages from the * list for the current discussion are: 009177.html 009268.html 0498.html 0849.html My setup is an Eicon Diva (HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2)

Hi, I'm having a problem with DTMF tones from my SIP client apparently crashing the chan_capi driver. However I'm not sure whether this is a bug or misconfiguration on my part: if I set "softdtmf=1" in /etc/asterisk/capi.conf the problem goes away. Does the AVM B1 not support DTMF detection? The set up I have is using latest CVS (3 days old) running RH8 on a 933MHz P3. SIP