similar to: Supervision Issue With Asterisk/Sipura/Talkn

I am attempting to write in incoming context for calls. 1. If the caller id is given and it is not black listed it will Playback a greeting and then right the phone or go to voicemail under busy or unavailable conditions 2. If no caller id is given, then Privacy Manager will ask for the number. I am testing 6145551212 to see if the black list will work 3. If a caller id is given, and it is

Could someone with inside knowledge comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use

I'm working on setting up a macro that will allow users to call their own DID number, and when they hear their voicemail greeting hit the * key and be prompted for their password to check vmail. For some reason though the background command isn't working as I'd expect it to: [macro-vmessage] exten => s,1,Answer exten =>

Sorry if this comes in twice. Wasn't subscribed first time :-( Anyone help me here...... It worked once :-( 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.

Well, I thought I had my problem solved, but it is acting up again. Hopefully this time I can provide enough information. What I have is an * box setup with one X100P and TDM400 with one FXO and one FXS. For my regular setup with interfacing with my PSTN and my entire house with analog phones, the box is working great. I am trying to interface a Mediatrix 1202 device to my * box via the

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

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

Hi list! I'm trying to get PrivacyManager working but for some reason it always thinks that CallerID is present (when it isn't). I get this on the console: == Primary D-Channel on span 1 up -- Accepting voice call from '' to '0711234567' on channel 0/2, span 1 -- Executing Ringing("Zap/2-1", "") in new stack -- Executing

I'm able to receive but not make calls with zaptel using an X101P connecting to Asterisk with an Xlite client. My client has context = flat in sip.conf and extensions number 8919 In extensions.conf I've got: [home] ; Line 1 ; exten => 8919,1,Dial(${PHONES1},20,Ttm) exten => 8919,2,Macro(vmessage,${PHONES1VM}) exten => 8919,3,Hangup [outgoing] exten =>

Hi, I am using asterisk 1.4.28 with freepbx and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. *My log file showing this while an incoming call on PSTN line:* tail -f /var/log/asterisk/full [Dec 30 06:36:16] DEBUG[2559] dsp.c: dsp busy pattern set to 0,0

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

Hi all, I'm looking for an easy way to lookup numbers from the database so I can fork calls from my daughters friends onto her IP phone/answering system. I'm looking for something very similar to LookupBlacklist, but I'm already using LookupBlackist to filter out telemarketers. What I'm doing now is adding multiple exten=> lines to my extensions.conf file to match those

Hi, I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is working fine except the caller ID of incoming call from PSTN line. The phone display is showing "Unknown" when there is an incoming call. I think the same problem listed here: https://issues.asterisk.org/view.php?id=6683 There is one patch on this link but i don't know how to apply patch on asterisknow.

I am new to Shorewall and FreeSwan, please excuse my ignorance I was wondering if someone could help me. I had help getting my FreeSwan running with the following iptables commands: iptables -I FORWARD -s 0/0 -d 192.168.1.0/24 -i ipsec0 -o eth1 -j ACCEPT iptables -I FORWARD -s 192.168.1.0/24 -d 0/0 -i eth1 -o ipsec0 -j ACCEPT If I manually run this FreeSwan works, however I am not sure

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

Hi folks - I've recently returned to asterisk after an eighteen month break. I've two sip providers - gradwell in the UK (inbound and outbound) and talklite in the US (outbound only). I've managed to get outbound dialing working but am not receiving any calls from gradwell. I've included my sip.conf and extensions.conf as well as the output from tethereal. When a call is placed

Hi, here is my scenario in my voip system(asterisk based) every user has a primary did and 5 secondary did's i.e. all six did's point to a single channel. every user has a blacklist feature and a call filter feature. if blacklist feature is enabled, user has option to include 5 bl numbers. if the user is using all 5 bl numbers then i have to match DNID with every(6) did. im not using

Mateo, Dialing the extension to your softphone is the same as any hardware extension. Exten => 1000,1,Dial,(SIP/1000,20,trf) pretty exten => 1000,2,Macro(vmessage,1000) exten => 1000,3,Hangup Change [mateo01] to [1000] in your sip and you will be saying that ext. 1000 is registered with the specifics you are using. Update the settings in your softphone to register the name and

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hello, I have a rather odd problem with Asterisk detecting faxes. I have two POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2 is fof fax. When I set them up with channel => 1-2 in zapata.conf, all is fine, but as soon as I have two channel => definitions, Asterisk is unable to detect faxes. The fax line is not supposed to ring local phones, so the most obvious