similar to: Need advice with "free line" notification

Barton Hodges wrote: > I've been messing around with a "free line" notification > where an extension is dialed for a second when a line becomes > available. I can't seem to get the "h" extension to continue > when the local party hangs up. I've seen references to other > people having the same problem in the list archives, and the > solution

In the following scenario, the user calling from a SIPphone registered phone is able to transfer the called user to another extension. sip.conf: [general] port = 5060 context = from-sip register => number:password@proxy01.sipphone.com extensions.conf: [from-sip] exten => s,1,Dial(SIP/111&SIP/117) exten => 111,1,Dial(SIP/111,20) exten => 117,1,Dial(SIP/117,20) 1. The calling user

We currently have a 15-phone system using Asterisk, a combination of analog phones/Grandstream HandyTone-286 and Grandstream BT101s, and 4 X100Ps connected to analog lines. The system works well except for the occasional echo problem. I have all the echo parameters configured, removed all the extra incoming analog lines except to the PBX, etc. following all the advice on the wiki and on the

Is anyone else experiencing problems with Transfer via # and the 'T' or 't' flags passed to Dial()? I've tried both the latest CVS and 0.7.1 tarball. If I dial in from a pstn line and then choose an extension that dials a SIP phone with "Ttm" flags, when I press # on the SIP phone, the pstn caller hears the "Transfer" and the SIP phone gets the music on

> Did you ever find out what was wrong? Nope, and personally I think there's a bug in 2.4.19. Since I sent the email below I did a make oldconfig and went thru the new options very carefully - no luck. I'm still running on 2.4.18, and I can compile a kernel, install, and it works fine. Randy Barton Hodges wrote: > Hi Randy > > Please pardon the intrusion, but I found your

Hi. There is not much info on the FWD site about this. What codecs do they use? When I try to connect with X-Lite, it works with GSM. When I try to call out with *, it wants G729. I have disallow=all and allow=gsm in the sip.conf. I end up getting errors: Unable to find a path from G729A to GSM Unable to find a path from GSM to G729A What's up with that? I was able to make a call once

Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). => When a call comes in, I'd like an AGI

Like many others on this list, I had been experiencing periodic lockups with my Grandstream products (Handytone 286 ATA & BudgeTone 101). The lockups consisted of seemingly dead devices, no dialtone or response, until I power cycled via software or hardware. The workaround had been to reboot the device every 30 minutes with a cron job. I contacted Grandstream and although they didn't

Hi folks, I'm wondering if it is currently possible to configure Asterisk to forward the conversations from 2 channels into a streaming daemon, such as Icecast, so that other people connected to the Internet can listen. The concept is similar to a radio talk-show. The show host would connect to Asterisk via an X100P or through VOIP. His or her voice would then be sent to the streaming

> I'm going blind searching - maybe you know? > > During the execution of a script I want to play fake ring to caller. > Both of these examples complain of missing option: > > $agi->exec("Ringing"); > $agi->exec("Playtones ring"); > > Notice: Undefined variable: options in > /var/lib/asterisk/agi-bin/includes/phpagi.php on line 326

Hi Folks, I''m using a 2.4.x kernel and TC from the iproute2 package so that I can limit traffic through my gateway. I''m using this to mark packets when they leave the LAN: /sbin/ipchains -A forward -j MASQ -i eth0 -s 192.168.1.0/24 -d 0.0.0.0/0 -m 1 When the packets return, I need to have them marked again so that the ingress filter will limit the bandwidth in the opposite

Hi; Before I tell a customer that this would require custom development I figured I would ask here. Does Asterisk support pager notification of new voicemails out of the box? Or do I need an AGI script to do that? Also, if I want to call a number from an automated program in Asterisk and get the DTMF tones entered by the user on the other side, is there an easy way to do this? Best

Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages? I tried to use the "minmessage", but, couldn't. Is

Short question: how to get control or notification (gosub, macro, AGI) when a queue member's phone starts ringing due to an incoming call from the queue. Backround: Our phone operators serve both an asterisk call queue and a queue for web chat support. I have a gosub on the queue that calls to our app server to mark the operator unavailable for web chat as soon as they answer an

Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Also, anyone have suggestion on licensing? LGPL? FreeBSD? Thanks

I would like to have my asterisk ring my cell phone and let me know when a new voicemail arrives. In fact if it would automatically put into the voicemail menu that would be cool too. In the future I will probably want it to IM me. Are there good examples somewhere of doing stuff automatically on the arrival of a new voicemail ? I noticed a place for the pager email address in voicemail.conf,

Attached is the log. I'm building with: clang version 10.0.0 (https://github.com/llvm/llvm-project/ 3a6da1122b990386edeba0987d0d1fdc9c8dc53d) Target: x86_64-unknown-linux-gnu Thread model: posix On some Ubuntu-like distribution. I also ran with ASAN once and it found a bunch of leaks in bin/tco. Best, -- Mehdi On Tue, Apr 7, 2020 at 4:36 AM Richard Barton <Richard.Barton at

Hello All, I am looking for an R library/function that allows the specification of a custom link function in a linear mixed model. I've been using glm() in library MASS to fit fixed-effect models with a custom link but my study design demands mixed models. Any suggestions on the best R library/function to achieve this would be greatly appreciated. I have tried, to no avail, to

Hi all, I have the below peace of my AGI script...the problem here is that I cannot fetch the extension value to inside the script and assign it to another variable...I highlighted it in red #!/usr/bin/perl #use DBD::mysql; use DBI; use DBD::mysql; use Asterisk::AGI; ############################ #To read asterisk variable values. $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse();

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If