similar to: Transfer Asterisk 1.6 Telephone IP

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to doing a transfer from an analog extension to a SIP extension but until the moment I was not successful. I was testing both the recall key and uncomment the following lines in the features.conf file: blindxfer => #1 atxfer => *2 verifying previously that the extension uses the arguments "tT" with the Dial

Hi ALL i implemented asterisk as my home PBX system my * machine recieves a call and transfer this to my computer The problem is this that i get my voice back mean there is too much echo(there is no complain from the caller). I have set following values in zapata.conf echocancellation=yes (also tried different powers of 2) echocancelwhenbridged=yes is there any other setting or not ??or this

https://bugzilla.mindrot.org/show_bug.cgi?id=2720 Bug ID: 2720 Summary: Include username in "Permission denied (publickey)." message Product: Portable OpenSSH Version: 7.2p2 Hardware: Other OS: Linux Status: NEW Severity: enhancement Priority: P5 Component:

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Hello, We would like to continue a Perl AGI after a Dial() it has done completes following caller hangup. We would like to do this in the same AGI, and not using a new AGI from the 'h' extension. It works fine when the called party hangs up and the 'g' option is used, but not for caller hangup. Is this possible? If not a confirmation that this is the case would be very helpful.

mdew:~# tc qdisc add dev imq0 handle 1: root htb default 12 r2q 1 Cannot find device "imq0" mdew:~# lsmod Module Size Used by Not tainted ipt_REDIRECT 728 0 (autoclean) ipt_MARK 728 2 (autoclean) iptable_mangle 2100 1 (autoclean) ipt_REJECT 2712 4 (autoclean) iptable_filter 1672 1 (autoclean)

Hello, We have a customer who upgraded from Asterisk 1.4 to 1.6.1.22 and is now experiencing higher CPU utilization on their server. I can't see anything wrong, so is this just expected with 1.6? Can anyone help explain it? Thanks for any advice. -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 8063 9019

Try to build samba-4.0.23 (4.0.22 build without problems): http://git.altlinux.org/tasks/136130/build/100/x86_64/log There is error as described in "new" bug https://bugzilla.samba.org/show_bug.cgi?id=9515 [2720/2814] Generating manpages/smb.conf.5 ... (many validity errors like ID ABORTSHUTDOWNSCRIPT already defined) ... runtime error: file

Dear R-list members, I am calculating the linear extrapolation for a data set, using the function found in Hmisc. x=c(0.0265,-0.0003,0.0142,0.0263,0.0634,0.1145,0.2504) y=c(58,107,152,239,362,512,724) x1=0.0393216 approxExtrap(x,y,x1, method="linear") approx(x,y,x1) #to see what is happening: plot(x, y, typ="o") abline(v=x1, col=8) Which gives x=0.03 and y=163, instead of

I am getting frequent segfaults on a new Asterisk installation. So far the only message I see is: Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 00007fb2d535723f sp 00007fb25a11b5c0 error 4 in libasteriskpj.so.2[7fb2d52e5000+180000] Jul 18 09:17:00 pbxbogota kernel: asterisk[27453]: segfault at 188 ip 00007f4afea0c23f sp 00007f4a7f7e35c0 error 4 in

Hello, We have a quadBRI in NT mode with bri_cpe_ptmp signalling and when connect a ISDN telephone to this nothings happen. What can I do? My config files are this: Zaptel.conf: loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,1,3,ccs,ami span=3,1,3,ccs,ami span=4,1,3,ccs,ami bchan=1,2 dchan=3

I connect Asterisk to my analog PBX using X100P. In my analog PBX, I need to dial 0 (zero) to pick up the line. How can I use Dial command to dial (0 + telephone number) directly? I used exten => 10,1,Answer() exten => 10,2,Dial(Zap/1/0) exten => 10,3,Hangup It works, but I need to dial 10 and after the ring tone, the telephone number How can I do?

I'm new to this list. Reading the asterisk handbook pdf (good work) but but still have some questions. Using Trustix 2.1 and installed Asterisk via CVS, zaptel and libpri. We have a dedicated server which is connected to our telephone company. It makes us able to call ordinary phones via VOIP using Ericsson DRG22. Would like to make people able to call me - and get a message "dial 1

Hello, I'm really new on Asterisk. Is it possible to use a telephone machine connected to a modem as an asterisk voice input output device? I do not need PSTN connection. The scheme i'm thinking about is; user -> phone -> modem -> asterisk -> ip -> vice versa. If it is possible can a user dial another asterisk user via the phone? I've searched astersik lists but

Hi, I have a Cisco 7960G phone for which I know the pinout of the headset socket. I have a couple of standard telephone headsets which I do not know the pinout of. I'd like to connect the two. If I have the pinout of a normal/standard headset I can rewire the ones I have to match the cisco. Thanks Mike

Hello All, I am looking for a IAX2 analog telephone adapter, just want to ask your views on which ones are bad, good and the best. Thanks in advance, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh@imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg

Hello I am using a Snom 190 and the quality seems OK. Trouble is the volume is quite low and full volume on the Snom is still too low. Is there something I can do on the asterisk to increase the volume? Angus

I have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com

I have an interesting task for my son's lacrosse team... it is the time-old telephone tree... I am pretty sure someone has already done this w/*, why re-invent the wheel?... a) coach calls in leaves a msg, others call in retrieve the msg b) coach calls in leaves a msg, kicks of a call to every parent plays msg c) coach calls in leaves a msg, kicks off a call to every parent, checks for

Hello everybody. I have two AT530 telephones and one X-Lite extension conected to my Asterisk. This is part of my extensions.con. exten => 105,1,Answer exten => 105,2,Background(/home/user/suport) exten => 1,1,Dial(SIP/101,30,Ttm) exten => 2,1,Dial(SIP/102,30,Ttm) When I call to 105 extension from the AT530 telephones and I select option "1" it doesn't redirect to