similar to: Line-in as MOH source

Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary

Hello All I am wondering is someone knows how to configure the * to work with an Ericsson MD-110 with SL60 signaling?? through a TLU76 card. What is the right configuration in the zaptel.conf ? I currently have it configured as span=3.0.0,ccs,hdb3,crc4 but it doesn't detect anything when I connect it to the PBX and no activity can be seen either in the logs or in the asterisk console.

Same problem here: if call come over ISDN PRI and it is for a SIP phone that equals to strong echo situation, at the SIP end. Interestingly this doesn't happen on all calls but it does on 95% of them. Asterisk load at that moment is insignificant - 1 to 2 calls. I have tried with all possible echo cancellers in zconfig.h, with and without MMX, and with and without CFLAGS+=-march=i686 in

Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? I know this has to work somehow. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/4a3c3025/attachment.htm

Hi list! I'm trying to configure the Kirk IP 600 wireless (DECT) system. The wireless phones are regular DECT phones, the Kirk IP 600 is doing the voip part by registering/gatewaying the phones to a callmanager server. The phones do not work and I think the problem is that they do not register at asterisk but I'm not sure because I don't understand a thing of the sccp/skinny

I thought this had been around before but I can't seem to find anything about it. I have a customer whom prior to upgrading to Asterisk invested in one of those boxes that plays your company sales campaign into the MOH port on your key system. For reasons of message maintenance he wants to keep the box as part of the new system. Can I couple this to the sound card in the Asterisk server

Is there a way to execute specific applications at Asterisk startup (like "startup context" or something)?

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

Hi, I have a SPA-3000 and would like to use the 911 recipe from http://www.voip-info.org/wiki-Asterisk+tips+911. So I took the simple recipe and modified it slightly: exten => 911,1,ChanIsAvail(SIP/potsoutbound) exten => 911,2,Dial(SIP/potsoutbound/911) exten => 911,3,Hangup() exten => 911,102,SoftHangup(SIP/potsoutbound) exten => 911,103,Wait(1) exten => 911,104,Goto(1) Now,

ok switching to "Console/default" does show the text --- <("<) --- Call to device 'default' on console from 'default' <2564286000> --- (>")> --- --- <("<) --- Auto-answered --- (>")> --- However I don't hear any audio. Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to use SoftHangup in my dialplan, but it's either not working or I'm not using it correctly. when i'm on the console, i see: pbx1*CLI> core show channels Channel Location State Application(Data) SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line)) SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,, 2 active

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

>> See if it plays back properly. > > Running aplay as asterisk user seems to be no problem: > > asterisk at puppy$ aplay /usr/share/sounds/alsa/Front_Center.wav > Playing: WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit > Little Endian, Rate: 48000 Hz, mono > asterisk at puppy:~$ aplay -Dpulse /usr/share/sounds/alsa/Front_Center.wav >

Hi, Is it possible to softhangup a channel based on priority. I mean I want to put some calls in higher priority lets say 100. If all channels are busy and somebody wants to dial an extension with priority higher than 100. How can softhangup drop a line which has priority less than 100? I will appreciate your valuable help. Thanks Smir

Looking for a radio tuner for my audio server. Needs to do AM. Any hardware reccomendations? -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Principal Consultant 10 West 24th Street #100 - - +1 (443) 269-1555 x333

Asterisk doesn't work with the latest CVS of ALSA (2003-06-24) . The module chan_alsa.so won't load even if the oss module, chan_oss.so, isn't loaded. There are no error messages. I've been chasing ALSA/Asterisk/client problems in one form or another for some time now. In previous versions of Asterisk and ALSA -- i.e., last week -- I could load either chan_oss.so or

I have just installed the Alsa drivers for my 2.4.18-14 kernel (RH8). I have configured the sound card ok with alsaconf and tested with the aplay , works fine. But when I run asterisk it says.. ------------------------------- [chan_alsa.so] => (ALSA Console Channel Driver) Apr 20 18:28:34 ERROR[8192]: chan_alsa.c:339 alsa_card_init: snd_pcm_open failed: No such device or address Apr 20

On Thu, Aug 30, 2012 at 6:33 PM, Konrad Rzeszutek Wilk <konrad@kernel.org>wrote: Hi, > > I''ve just upgraded a server of mine from a Core i3 2100T to an i7 3770, > in order > > to do full virtualization with VTd. > > > > I''m using kernel 3.5.2 and Xen from git://xenbits.xen.org/xen.git @ > commit > >

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten =>

Hi, has anyone tried to use wine to control a tv tuner card ? any hints tips advice gratefully received Richard