similar to: Voicemail not working with vm boxes named with a star

Hi I installed Asterisk 1.8.7 with CD ISO(Elastix 2.2) I want to use DAHDISendCallreroutingFacility Application on a PRI link(LIBPRI Already installed). according to https://wiki.asterisk.org/wiki/display/AST/New+in+1.8 Asterisk 1.8 include this application but I cannot see it with "core show applications" Do I need to install mISDN or other modules for using that ? Regards M.Shirazi

I am having issues with Blind Transfer on asterisk 1.8 If I call from one Grandstream phone to another and us the transfer key to do a blind transfer everything works fine. When calling in on a sip trunk and then trying to use the transfer key to transfer from Grandstream phone to Grandstream phone the call just hangs up. It did not do this on Asterisk 1.4.x or 1.6.2.x . If we use

Karsten Wemheuer <kwem at gmx.de> schrieb: Hi Karsten! > the timeout value of 15 minutes directs me to an issue with session > timer. Try to refuse them by putting the line > session-timers = refuse > into the general context of sip.conf. Reload the sip stack with "sip > reload". Sorry, I forgot to mention that... I already have this setting:

Hello all, I've been lobbying for some time at the #asterisk IRC channel. Until now, I still can't find a solution to my one way audio problem. I rebuilt the Asterisk-1.4.21.2 from the Debian Testing repository on my Debian Etch. I got a Digium TDM400P with 1 FXO (channel 4) and 1 FXS (channel 1). My SIP extension phone located inside the LAN is a SNOM 300 IP phone. This one way audio

Hi All; Already in the voicemail.conf file, I added the extension 500 and kindly find below my voicemail configuration: [Internal] 0 => 1234,Gama Operator,Operator at gama.com 500 => 1234,Operator,Operator at gama.com 501 => 1234,Employer Name,employer_email at gama.com 502 => 1234,Employer Name,employer_email at gama.com Asterisk version is 1.8 and currently I am getting this

Hello steve. Here is a patch I wrote for app_voicemail.c which does exactly as you describe. When the outgoing message is playing, if the listener hits the "*" key, they're prompted for a mailbox and password, whereupon they can check their voicemail as if they were using the internal phone. I found no other way of doing this. If you patch your app_voicemail.c, I have V1.44 from

Hi list! My Problem: all calls to international numbers will be dropped after exactly 15 minutes... I have a VoIP-account by Deutsche Telekom. This is what I see when I call someone (my parents) and the connection will be dropped: == Using SIP RTP CoS mark 5 -- Executing [+39015222222 at default:1] Set("SIP/00493511111111-00000125", "newNumber=0039015222222") in new

Hi All I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don?t in the initialization... I have in my file that comands: touch /var/lock/subsys/local modprobe zaptel modprobe wcfxo safe_asterisk I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening Somebody

I (sometimes) get this error message: WARNING[17191]: app_voicemail.c:2411 leave_voicemail: No entry in voicemail config file for 'irstname.lastname' I can see the value of the argument is "firstname.lastname" when this line executes in the std-exten macro: exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable... But the error message drops the first character. It

hi all, I can not get voicemail working in realtime with asterisk-1.2.6. extconfig.conf is correct voicemail => odbc,asterisk,voicemail_users i am getting the fallowing error Executing Answer("SIP/xx.xx.xx.xxx-0a02e1c0", "") in new stack -- Executing Set("SIP/xx.xx.xxx-0a02e1c0", "foo=102") in new stack -- Executing

Hi I do have asterisk running in real time I do want to add voicemail to real time. I did follow : http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail However when I do try to make a voicemail I do get : [Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible codecs, not accepting this offer! -- Executing [999alijawad at a2billing:1]

Hello, We recently upgraded our Asterisk box from 1.4 to 1.6.1. In both versions of voicemail you can press 3 for advanced options, 5 to leave a message and enter an extension to leave a voicemail. This feature worked fine under 1.4. Now under 1.6.1 all the prompts are the same but when you enter the extension it reads back the extension (or says the recorded name if present) then goes straight

Hi, I have problems with SIP via TLS. Asterisk works as a client. The TCP connection is established, followed by a client hello from Asterisk to the server. The server sends Server Hello, Certificate, Server Key Exchange and Server Hello Done. Than Asterisk sends back a Alert (Level: Fatal, Description Handshake Failure). The following line appears in the log: ast_iostream_start_tls: Problem

Hi, I input "0203#" after "mailbox?" voice prompt from Voicemail cmd on extensions.conf such as exten => 0021,1,Ringing exten => 0021,2,Wait(1) exten => 0021,3,Voicemail exten => 0021,4,Hangup *CLI> -- Executing [0021 at sip:1] Ringing("SIP/0103-09a308b0", "") in new stack -- Executing [0021 at sip:2]

Hi, I've installed * 1.2.1 with latest bristuff patches (0.3.0-PRE-1f). When I activate music-on-hold on a SIP-to-SIP connection, the music sounds like in a fast-forward play mode. On the *-console I can see much lines like this: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=18) What's going on? With bristuff 0.3.0-PRE-1d everything works fine (but

Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with "j" letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of

I've the following dialplan. exten => _XX,hint,SIP/${EXTEN} exten => _XX,1,Dial(SIP/${EXTEN},10,j) exten => _XX,2,VoiceMail(${EXTEN}@default,u|j) exten => _XX,3,Hangup() exten => _XX,102,Goto(110) exten => _XX,103,Playback(pbx-invalid) exten => _XX,104,Hangup() exten => _XX,110,VoiceMail(${EXTEN}@default,b|j) exten => _XX,111,Hangup() exten =>

Have a fair amount of asterisk experience, but this one is blowing my mind.. I have a context setup as follows: [department-listing] exten => s,1,Background(custom/6000) exten => s,2,DigitTimeout,5 exten => s,3,ResponseTimeout,30 ; exten => 1,1,Answer exten => 1,2,Wait(1) exten => 1,3,Background(pls-wait-connect-call) exten => 1,4,VoiceMail(u620&122) exten =>

Hi all, I have been trying to get my asterisk (v1.2.10) to lookup voicemail config data from my mysql database as opposed to voicemail.conf + sip.conf for my users. Users register with SER and get passed through to asterisk when they dial out. I followed the instructions as per http://www.voip-info.org/wiki/view/Asterisk+voicemail+database so basically I have 1) Build asterisk-addons-1.2.5 and

Am 17.06.2020 14:37, schrieb Karsten Wemheuer: Hi Karsten! > The product is "All-IP" and not the SIP trunk, right? > The call starts normally and after about 15 minutes the quality is > disturbed? No, current we have Magenta Zuhause. Tomorrow we'll change to DeutschlandLAN IP (business contract). The quality is disturbed from the first second... I had the problem, that