similar to: SetMusicOnHold

Hello, I'm trying to do something really easy : transfer a PSTN call to a H323 client. This works great. Now I'm trying to use the SetMusicOnHold function. I din't find any doc about it, I've just seen some mails in the list archive, but it still doesn't work. That's my extension.conf : [incoming] exten => s,1,SetMusicOnHold,default exten =>

Hi all, Can any one please help me in intergrating PHP/Mysql with my running asterisk server to configure IAX or SIP users? I will highly appreciate any help in this regard. Regards Nawaz. --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, >

Here is my scenario: Sip phone number 1 and 2 are defined in sip.conf, and both have musiconhold=<class> set to the same outbound class that I want. This works fine for outbound calls (out to the pstn) Also, in extensions.conf for each extension that is setup to dial each of those sip phones, the first priority is SetMusicOnHold(<class>) So this works when a call comes in from the

Hi, Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram. For some odd reason now that I have the asterisk box almost to the stage I want it, I hit a problem. I have a te405p in the system, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the

Hi, Urgently response would be wonderful, system is a Fedora Core 2. I have a Ericsson BP250 connected to 1 port on the TE405P and another connected to a local telco ISDN30. I have been running CVS-HEAD from about a 2 months ago and upgraded it again just in cause it was a version issue (didn't fix it) but this is what I am getting. When a person calls out from an extension on the BP250 to

Typical user error, one user forwards his calls to another using CFwdAll on Cisco 7940, but the user receiving the call has done the reverse. -- Called 117 -- Got SIP response 302 "Moved Temporarily" back from 10.139.2.15 -- Now forwarding Local/117@sip-0eb7,2 to 'Local/114@sip' (thanks to SIP/117-df17) -- Executing Macro("Local/114@sip-35cf,2",

I've been trying to play the default music on hold file, but no luck yet. here is my configuration: extensions.conf [incoming] exten => s,1,Dial,Zap/2|10 exten => s,2,Voicemail,u34 exten => s,102,Voicemail,b34 exten => 34,1,SetMusicOnHold,default Musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random =>

Hi Everyone, Merry Christmas :-).... My Asterisk Box doesn't have a sound card, it is running Asterisk 1.02 Zaptel 1.02 Libpri 1.02 Mpg123 0.59r All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2 Any help would be very much appreciated..... The error I am getting is -- Executing WaitMusicOnHold("SIP/snom-james-849d", "30") in new stack Dec 12 00:27:29

I have a rather long dial plan, but it includes support for call waiting. However, the setgroup checkgroup commands don't seem to be working. Can anyone help on this one? Excerpts are below. First exten-vm is dialed and then dial-new. As I understand, priority 1 increments the active channels for the caller and then in "dial-new" priority 8 increments for Arg3, or the Callee

I've been banging my head against a brick wall for the last hour and I'm sure this is one of those easy to solve things - just that I can't see the wood for the trees. I'm trying to do: ----------- [some-context] Exten => s,1,Macro(dodial,'SIP/201,15,tT',123456,MOHClass) [macro-dodial] Exten => s,1,SetCallerID(${ARG2}) Exten => s,2,SetMusicOnHold(${ARG3}) Exten

Charles S. Antrim wrote: > I am using a card that has an fxo and fxs module. I am no where near an expert but I have my sip phone working through my pstn line and this is my config. /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 192.168.69.1 context = sip disallow = gsm allow = alaw disallow = ulaw nat=disable srvlookup=no localnet=192.168.69.0/255.255.255.0 subscribecontext =

Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. On ext 690, button 1 is setup for ext 691, I did this using both methods 691, and <sip:691@192.168.69.1> On ext 691, button 1 is setup for ext 690, I did this using both methods 690, and

I am trying to find an example of how to set up my FXS Station Card in my Asterisk. I have (1) XP100P I have (1) tdm20B (2 Port FXS) Could someone tell me if this is correct? /etc/zaptel.conf fxsks=1 fxoks=2 fxoks=3 loadzone=us defaultzone=us /etc/asterisk/zapata.conf [channels] ; language=en ; ;X100P Port 1 context=inbound-analog signalling=fxs_ks usecallerid=yes echocancel=yes

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

After a long working evening yesterday, now my * box place and receive calls with H323,SIP and ISDN line. Calling from the office to an outside line, happens: - If I call a mobile number and the called answers, all goes ok - If I call a number at home/office, and it's answered , all goes ok - If I call a toll-free number with an IVR system, nothing happens: it continues to ring indefinitely

Hello... I have a big problem with asterisk. Every time i make a call asterisk does not bridge the zap channels. The zap channel from which i'm calling remains in state:ring and applicaton:dial and the zap channel with the external line configured remains in state:dialling an Application:AppDial. Zap/20-1 agentie s 1 Dialing AppDial (Outgoing Line) 09399 (None) Zap/9-1 int_omg 09399 5 Ring

I posted a message a little while ago but got no response (that I can recall), I've also seen other people mention this issue. Basically, when you have music on hold, it doesn't play the music on hold, the debug info shows it is starting and then stops straight away.. # My extensions.conf ... exten => s,1,Answer exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,10 exten

Dear all. I have what appears to be a configuration error but I cannot for the life of me see what it is. (I am a newbie) I have searched the wikki and google etc but still none the wiser. Any help would be very gratefully received. Problem: Unable to make outgoing calls via E1 euroISDN Digium TE110p card, given congestion signal as per config, unable to open zap channel. All incoming calls work

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

Ok, I'm tearing my hair out trying to get Asterisk moh streaming to work. After several hours jerking around with icecast and muse, I tried to point my asterisk system directly at two streams I know work. This is what extensions.conf has: [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 [stream2] mode=custom directory=/var/lib/asterisk/mohmp3-empty