similar to: oh323 and Asterisk: Calls always hang up

Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring.

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

Hi, We have a system that recorded voicemail for about an hour after the caller hungup. I'm going to put a timeout on it but is there anything to look for that can help prevent this? The system is running on a telenet line in Belgium. The answer dialplan I used was: [macro-stddial] exten => s,1,Answer exten => s,2,Playback(transfer) exten => s,3,Dial(${ARG2},60) exten =>

I'm having a bit of a problem here: I have a * box with a fritz isdn card (running capi 2.0 and chan_capi) and a x100p card for testing purposes. As a proof of concept, I wanted to be able to dial into the * using the isdn line, listen to a message, and enter a 3 digit extension number. If this happens, I wanted the * box to dial out using the x100p card, into our PBX (Nortel Meridian). If

This is the core trace (gdb) bt #0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6 #1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828 "default", interpclass=0x0) at res_musiconhold.c:646 #2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 <Address 0x64 out of bounds>, interpclass=0x88 <Address 0x88 out of bounds>) at channel.c:4614 #3

Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch

Hi there, I'll need some help with this: Trying to establish an IAX2 link between two servers works in one direction (SIP client with ulaw), but not in the other (SIP client with GSM). The client used for this is X-Lite behind NAT while both servers have a public IP (I playback an anouncement before trying to connect to the second *). Error on the originating * server:

Hi List, I'm working on making one of my applications multi-lingual and find that I have this problem. The SayDigits and SayNumber functions in 1.4.26.2 recognize but don't process the CHANNEL(language) function. Here's a snippet to verify. exten => 317,1,Answer exten => 317,n,playback(tt-monkeysintro) exten => 317,n,Set(CHANNEL(language)=es) exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I try to get callers forwarded to by mobile phone when they dial a certain digit. In my extensions.conf I have defined the following: [279xxxx] exten => s,1,SetLanguage(de) exten => s,2,Wait,5 exten => s,3,BackGround(demo-congrats) exten => s,4,Goto(boksa,#,1) exten => 3,1,VoiceMail,u1 exten => 4,1,VoicemailMain exten =>

Hi! I'm here again with echo canceller problem... :-( I think I've done everything to enable echo canceller feature, but it still doesn't work... Can anybody tell me if there is some error or something missing in this configuration please? I'm using Eicon Diva Server 4Bri. http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm?techspec=1&regID=4999 Card

I finally got Solaris to successfully make asterisk, using these instructions: http://sunfreeware.com/programlistsparc10.html#gcc33 Now though, when I issue the make install, I get this error: mkdir -p /var/opt/asterisk/spool/system mkdir -p /var/opt/asterisk/spool/tmp mkdir -p /var/opt/asterisk/spool/meetme install -m 755 asterisk /opt/asterisk/usr/sbin/ install: asterisk was not found

Hi everybody, I am having a problem with * Record Application. The thing is I don't want the "beep" before recording, so I removed the instructions: ast_streamfile(chan, "null", chan->language); ast_waitstream(chan, ""); ast_stopstream(chan); Now I am having a strange problem. After I record the sound, the recorded file gets a 3 second of silence before

Followed this; #cd /usr/src #export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot #cvs login (password is anoncvs) #cvs checkout zaptel libpri asterisk #cd zaptel ; zaptel equipment #make clean; make install #cd ../libpri ; isdn #make clean; make install #cd ../asterisk #make clean ..but

Hi, today i have taken a strong look at meetme.c what i am trying to accomplish is the following: it should be possible to access an menu from within the conference in order to perform special tasks, eg. to dial another number so that the called person is joined with the conderence. my first try was to use an agi-script for this, but as with agi enabled sip-channels (for which

Hi, I have a question about this here: Asterisk-Version: 11.10.2 File: res/res_agi.c Line: 2377 [...] static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[]) 2304 { 2305 struct ast_filestream *fs; 2306 struct ast_frame *f; 2307 struct timeval start; 2308 long sample_offset = 0; 2309 int res = 0; 2310

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

I am seeing an issue with getting certain sip devices to be recognized as defined SIP clients host= in the sip.conf and the only deference that I can find btw sources that work and don't work is that devices that send packets with an Initial Via header of themselves appears to work and pick the context correctly but those that don't have the Via just get dropped in the context of the

Hi chan_capi users, this thing is awesome, no delays like in modem_i4l! Plus, it got those nice ISDN features. Here's my question: Does my service provider (Deutsche Telekom) have to provide me with these Services (CD, ECT)? (the Readme in 0.2.5 says "does not relay on service CD") I know, that I don't have CFU,CFNR,CFBS (which I would have to order seperately). How likely

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

Hello I need to write a script that will dial a list of customers and play a message. I couldn't find a way to tell Asterisk/Zaptel to wait until the callee has actually picked up the phone before proceeding with Playback(): ============ ;call made through Dial(): Doesn't proceed after off-hook/hangup [internal] exten => 8888,1,Dial(Zap/1/${IPPI}) exten => 8888,n,NoOp(We never