similar to: Background() command

Hi all. I am new to the list and new to asterisk. I have asterisk installed and running. I am using it as a voicemail server only. What I would like to do is send users to a general mailbox that will be addressed as <companyname>@asterisk and give them the option to wait for the tone and leave a message, or press 9 to dial by name. My questions are: 1. What is the best way to do

I am setting up my * for at home office and still have analog phones attached and answer from those analog phones and not necessarily through the pbx. I found that with the X100P cards, they see the 2nd ring and will be ready to answer the line. I used a Wait to pause and allow another 2 rings before * answers. But found that if we answer the line after the 2nd ring and before the 4th, * still

I want to limit the number of simultaneous incoming calls that my IAX DID can accept to, say, 2. The IAX DID provider sets no limit. The code below does work, but when the limit is in effect, new callers hear a "call cannot be completed as dialed.." message instead of a busy signal. Maybe this is an issue with the provider, but I do not like this and want callers to hear a busy signal.

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

IPSwitchBoard Version 0.69 has just been released; it is available for FREE: <http://mambo.thorben.dk> Download here Release notes: * Record calls by right clicking any extension button, you can have several recordings at the same time. * Bug fixes The recordings will be placed as a single wav file on the Asterisk server in the folder: /var/spool/asterisk/monitor the name of the

I just tried this in my extensions.conf exten => **,1,Noop(Testing) exten => **,n,Playback(demo-congrats) Did a reload... and the above does not happen. I created as 12 instead of the ** and that works fine. Is there anyway to get the ** to work? I also am using a polycom phone if that affects things. I'm using asterisk 13.15.0 Thanks Jerry -------------- next part --------------

Is there a way to pass variables/arguments to the h extension ? for example : [default] exten => _1098933X.,1,NoOp(CARRIER TWT->TIM, EXTEN: ${EXTEN}}, SIPCALLID: ${SIPCALLID}, SIPDOMAIN: ${SIPDOMAIN}) exten => _1098933X.,2,SetVar(_PROVA="bla") [lot of stuff, agi, goto, tricks and magic that happens] exten => _1098933X.,10,Dial(${CHAN_DEST},,L(3600000:3599900)) <-

Hey guys, Cal someone help me. I'm register voiceplus DID i try to config fllow example but not work. When i test call to number and debug iax2 in my asterisk not found packet. My iax.conf -------- register => in-xxx:yyy@gw5.voicepulse.com [voicepulse] context = voicepulse-incoming secret=yyy auth=md5 type=friend host=gw5.voicepulse.com ------ extention.conf ---- [voicepulse-incoming]

Dear Members, I am trying to make call from one Gnophone to another through the local Asterisk Server.All the three systems have local IP Addresses I created two users "sheeba" (extension 600) and "test" (extension 602) in iax.conf file: [sheeba] type=friend auth=plaintext host=dynamic secret=sheeba context=default allow=gsm permit=0.0.0.0/0.0.0.0

Hey hey... I'd like to create a new feature code in asterisk so when a user dials... say... *00, it would then call some other extensions and play a sound file to them. So far, this is what I have... [testing-custom] exten => *00,1,Wait(1) exten => *00,2,Playback(beep) exten => *00,3,Playback(beep) exten => *00,4,AGI(festival-script.pl|I will now attempt the call) exten =>

I have the following in the dialplan: [macro-systemrecording] exten => s,1,Goto(${ARG1},1) exten => dorecord,1,Record(/tmp/${CALLERID(number)}-ivrrecording:wav) exten => dorecord,n,Wait(1) exten => dorecord,n,Goto(confmenu,1) exten => docheck,1,Playback(/tmp/${CALLERID(number)}-ivrrecording) exten => docheck,n,Wait(1) exten => docheck,n,Goto(confmenu,1) exten =>

in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here

Hi, I'm trying to call a SIP server while providing the SIP server username/password in dial string but it's not working ... Dial(SIP/gs102:test at 192.168.2.81); User on sip server (192.168.2.81): [gs102] disallow=all allow=ulaw allow=alaw type=friend username=gs102 secret=test host=dynamic dtmfmode=inband defaultip=192.168.2.1 qualify=1000 mailbox=102 context=context-gs102

Will Wait(n) still listen for DTMF input from the caller after there has been a Background(some-message) prompt, or do I need to use Background(silence/n) to still listen for DTMF? -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a

I've wikied and googled, but could not find any appropriate scripts - : I was wanting to stress-test my new server, and as I have a TE410p card (but only using 2 ports), I was going to connect ports 3 & 4 with a cross-over cable so that I could make a number of outbound calls on port 3 and receive them as inbound on port 4. I was also wanting to vary the destination of the outbound,

Hi all, I have setup my Cisco 79XX phone. Did the tftp, put the config files in the right location with the right names. Booted my phone, it does the tftp things, the screen shows my extensions everything seems fine. However, when I come offhook and try to dial 11 which is just a playback of demo-congrats in the dialplan the phone says Calling Out (INV) below is my sip.conf file - I presume it

my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go to 1.6 voice mail. it says == Using SIP RTP CoS mark 5 -- Executing [4567 at sip:1] Dial("SIP/312-09f9a720", "IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack -- Called trunk10 at 147.120.203.98/4567 [Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call rejected by

Hi, I've been working on starting a lab of end to end asterisk system and now most of pieces seem to be working. The two asterisk servers are connected by T1. Both servers have a couple of SIP phones connected and one of the servers has a FXS card with an analog phone hanging. I can make calls across the T1 link however there is one thing that I don't understand. I need to append one

Anyone help me here? I am a newbie so be gentle ;-)...... It worked once and then I played with the configs. I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22 I have the 7690 with a SIP iamge (Whatever latest is ) I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here

A while back, I posted about detecting when a call was picked up by a fax machine. It was suggested that having a "fax" extension and "faxdetect=yes" would cause it to jump to the "fax" extension. This was not something I could get to work. I have now created a very simple example. In sip.conf I have "faxdetect = yes". My example extension is: [test]