similar to: New CVS version

I search a ActiveX to develop one softphone SIP with codec G723. Who can help me? Thank?s Jo?o Carlos Moura

My TE110P reset some times in the day. E this cause an interruption in the service. How I decide this problem? my zaptel.conf span=1,1,0,esf,b8zs bchan=1-23 # set this to 1-15,17-31 for E1 dchan=24 # set this to 16 for E1 defaultzone=us loadzone=us my zapata.conf [channels] language=en signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=200 ; Asterisk

Hello, I was running a very simple test setup with * HEAD 7/15/2004 on Fedora Core 2 and things were working fine. Today I upgraded to RC1 and my asterisk service will no longer start. I downloaded the tarball, extracted, ran 'make', ran 'service asterisk stop', ran 'make install', removed all files in /etc/asterisk, ran 'make samples' and then ran 'service

Hello list I don't get to compile h323. I have the mistake: asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: ** [asteriskaudio.o] Erro 1 make[1]:

I am trying to setup a couple of virtual pbx's off of my one may asterisk box. So far I have been able to segment most everything via the Dial plan. My only question/problem has to do with the # Transfer function. I had set up # Transfers prior to segmenting the dial plan, and I cannot remember how I was able to specify which context to use when the user presses #. I haven't been able

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi i've installed asterisk by last cvs and i note res_parking.c is not anymore there; chan_capi-0.3.4b INSTALL file require: in /etc/asterisk/modules.conf insert the line: load => res_parking.so load => chan_capi.so running asterisk i get: [app_capiCD.so]Jul 21 15:32:26 WARNING[1076988448]: loader.c:242 ast_load_resource:

I attempted to install an X100P card but it was not correctly recognized by my Redhat 9 install. I had a test install running without any cards which was working great minus the outward dialing since no cards existed. Now that I have a card, I want to add it to the system. Do I have to scratch the whole current install in order to get the X100P running on my system or is there a way to get it

I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from

That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause

Hi, I am running Asterisk on Fedora Core 3. I am trying to use DDD to debug Asterisk. However, when I attach the debugger to the Asterisk Process, the Asterisk CLI promt hangs. Does not give any output, and Asterisk stops processing calls... What could be wrong and what is the best way to debug Asterisk...? Appreciate pointers.. Thx a lot, J --------------------------------- Do you

Does anyone have an update patch file to get Spandsp installed? I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 I installed spandsp-0.0.2 when runnig the patch I get patching file Makefile Hunk #1 FAILED at 41. Hunk #2 FAILED at 69. 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej -------------- next part -------------- An HTML attachment was scrubbed...

Hello, I recently tried an upgrade of CVS on my test server today and found that the res/res_parking.c file is completely gone. This is where I had to go into the code every time I do an upgrade and change the code to allow for doublehash transfers instead of single hash transfers: That means that you need to hit the pound key twice to initiate a transfer instead of once. Because of our inbound

Hi all, use the brought up to date version of the Asterisk and I have the following problem: Mine asterisk stop to register the extensions and I do not obtain to execute the command Stop Now. I do not see no message of error in logs. Somebody can help me? Thank's, JMoura

Anyone knows where can I find spandsp? Official site seems permanently down... TIA, Simone.

Hi, I'm trying to access a remote fiscal printer under Wine. I'm wondering if it's possible to use the 'net use' command from Windows (but under wine) to map the remote printer or if it's possible to simple mount the remote printer (dunno how as mount.cifs gives no clue about how to mount print shares) under 'dosdevices/unc' anyone ? Thanks in advance. Ricardo

hello I'm trying to figure out if anyone's accomplished putting someone on hold with a hardphone that doesn't have a hold button or multiple lines. I'm thinking transferring the caller to a specific extension or something...is this possible? Has it been done? thanks hank

Hello group I'm working on getting festival installed and working on my FC1. I ran into a problem and after searching Google I found this message talking about a patch for Speech Tools and Festival http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html The above site does not have the files. Does anyone in the group have this patch? Marc Sutter & Reed Wade do you still

Hello, First, sorry for my english. I'm a french student. I have a problem with asterisk. I use Budgetone SIP phones. When I dial 555 (VoicemailMain), I hear "You have 5 new messages, 1- Read your messages, 2- , etc ... ) But when I dial 1 or 2 or everything else, nothing happen. Are they some lines wich do that asterisk listen my phones ? Thanks for your help, have a nice day Thomas

Using CVS from 7/12/04 and trying to get the voicemail broadcast feature to work. Voicemail.conf has [mycontext] 3722 => 1234,BroadCast Test,,,cc=*@mycontext . then many other voicemail boxes. ----- whenever I leave voicemail at box 3722, only box 3722 gets the voicemail. It is not expanding it to other voicemail boxes in the [mycontext] context. Even if I replace the cc= line with