similar to: Playing a file upon pickup (dial command?)

Hi, I'm trying to do the following but can't quite get it right: 1) Callers rings DID number 2) Asterisk rings the appropriate channel for 30 second, if no answer sends to voicemail (no problem up to here, of course) 3) IF the channel is answered Asterisk plays an audio file 4) Asterisk connects caller with me I need to do this to "cover up" the delay within the first few

Hi all, i;m obviously a newbie, its been 2 days that im trying to figure out a way to deny a specific extension (300) from calling another specific extensions (03) except if the caller punch a specified password.. sorry if im not explaining myself well.. heres an example: i called my pstn line(with 300 as its sip account), an attendant answers and asks me to punch in an extension number right

CVS Head from 07/07/2005 I'm trying to make an IP-501 auto answer a call. exten => 301,1,SetVar(_ALERT_INFO="Ring_Ans") exten => 301,2,SetVar(ALERT_INFO="Ring_Ans") # Tried both combinations exten => 301,3,Dial(SIP/5001,15) exten => 301,4,Hangup Sip.cfg for Polycom phone <alertInfo voIpProt.SIP.alertInfo.2.value="Ring_Ans"

Hi to all the * people out there, Please kind to me as I am both new to Asterisk and to Linux - But I am learning fast. My config is quite simple, I'm just following examples and the Wiki: I have two PC's running X-Lite phones, these work without problems between each other, and I have a GS BudgeTone-100 registered to Free World Dial UP (working no problem). I have tried to

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hi, I have been having a problem with asterisk crashing when using local channels and realtime on asterisk 1.8.3-rc2. The example given here is I think the easiest way to reproduce this problem. In extensions.conf I have: [internal] switch => Realtime/extensions/p exten => 301,1,Answer() exten => 301,2,Dial(Local/501 at internal) exten => 301,3,Hangup() exten => 501,1,Answer()

Everyone, Im sure you've seen this error a million times, but Ive looked everywhere I can think of & still haven't found a solution that works. I'm trying to make an outside call, I can call the physical phone from a xlite on another pc (and vice versa) but whenever I try to make a call to the outside world, this happens: on the CLI: Jun 15 08:45:20 NOTICE[10390]:

Hello, I searched a while to find out, if there is an sftp-server implementation which provides an option similar to the defaultroot of proftpd. A typical use would be: DefaultRoot = ~ The option does the follwing: Once the use logs in, it determines the home directory of the user .ie /home/u1234 and takes this as the users root. The user cannot escape that root (he can not look at /tmp

Hi, I've created a test at "extensions.conf" like this with 3 steps: ; When dial 5555, get the first available channel and dial do 482343400 exten => 5555,1,Dial(Zap/g1/482343400,5,rt) ; When dial 5555, get the channel 20 and dial do 482343400 exten => 5555,2,Dial(Zap/20/482343400) ; Go to Voicemail 1234 exten => 5555,3,Voicemail(u1234) I've tried using just the

I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. Thanks, Brian

Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM

I've been using * at home for a while now and I'm quite happy with how it works. Having voicemail emailed to me and notify my cell phone via SMS is a great way to impress my friends. :-) The inbound context for my X101P looks something like this: exten => s,1,Dial(SIP/analog1&SIP/analog2,20) exten => s,2,Answer exten => s,3,Voicemail(u1234) exten => s,4,Hangup The

Greetings Friends, I would be grateful if you can help me undestand how to make my R code more efficiently. I have read in R intoductory tutorial that a for loop is not used so ofter (and is not maybe not that efficient) compared to other languages. So I am trying to build understanding how to get the equivalent of a for loop using more R-oriented thinking. If I got it right one way to do that

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

Hi to all! I have this config, PSTN <--> AS5300 <--> ASTERISK I am using the Cisco as5300 to receive incoming calls and routing them to Asterisk for IVR. When I ran asterisk this is what I get when calling the voicemail demo. *CLI> -- Executing Playback("SIP/-081058b8", "transfer|skip") in new stack -- Executing Macro("SIP/-081058b8",

Dear friends, please excuse me if my question will be trivial. I've installed and started Asterisk (stable 1.0.7, but with CVS HEAD I experienced just the same problem), and changed a bit sip.conf: [general] ; ... dtmfmode = inband disallow = all allow = ulaw allow = alaw allow = gsm run kphone, and call the 1235 extension. According to sample extensions.conf, Asterisk would

Hi All, I'm having trouble setting up a queue: I'm using AgentCallBackLogin to login in the queue, but: 1 - When an agent answer the call and another call arrive his phone rings again. 2 - When no there are no one answer the queue the system goes to voicemail of agent 1234 I'm using asterisk-1.2.0-beta1. My configuration is below, Any ideas? Many thanks, Joao Antunes

When releasing a virtio device, We can't free a struct virtio_device until the underlying struct device has been released, which might not happen immediately on device_unregister() even if that was the device's last reference. Instead, free the memory only once we know the device is gone in the release callback. Signed-off-by: Sasha Levin <sasha.levin at oracle.com> ---

When releasing a virtio device, We can't free a struct virtio_device until the underlying struct device has been released, which might not happen immediately on device_unregister() even if that was the device's last reference. Instead, free the memory only once we know the device is gone in the release callback. Signed-off-by: Sasha Levin <sasha.levin at oracle.com> ---

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List, I'm using the apllication AgentCallBackLogin so agents can login to a queue. They just need to enter the password, the CallBack Extensions is the ${CALLERIDNUM} Is there a way to AgentsLogOff withou using the AgentCallBackLogin application. I don't want the user to enter they CALLERIDNUM. Regards -----BEGIN PGP SIGNATURE-----