similar to: Stuck Parked Calls?

I have Asterisk running with a combination of SIP and H323 clients. I am using the OH323 module instead of the H323 one. When the SIP clients ring each other, they can hear a ringing noise in the ear peice to let them know that the other parties phone is ringing. However, when the H323 client rings a SIP client, there is no ringing sound at all, although as soon as the called party picks up the

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

Hi, I try to dial out through a Provider, but for that i need to be authenticated - it actually does not work !. What kind of dial command do i have to use when i: - Want to authenticate the user - And dial with that authenticated user an extension What i tried so far exten => _7.,2,Dial(SIP/USERNAME:PASSWORD@PROVIDER.COM/${EXTEN:1}) what have i done wrong ? i always get back a Service

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

Hello, I created a new field named pre_dst of type varchar(80) exactly like dst field in cdr table. In the dialplan I put: exten => _7.,1,Set(CDR(pre_dst)=${EXTEN:1}) and when I call, all goes fine except that pre_dst has always NULL value in cdr. Do you know why? Is something wrong I did? I know that original fields in cdr are only readable, but in this cas pre_dst is one I created

Possible Mediatrix 1204 fxo sip gateway workaround Need some feedback from experienced * users relative to this workaround please please please. Problem: The mediatrix 4-port fxo gateway does not provide any mechanism for * to select which "port" an outbound pstn call will use. (See lots of previous posts over the past four days for more detail if needed.) Our reseller has been

My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP phone. Here is my sip.conf file: ; ; SIP Configuration ; [general] context=default ; Default context for incoming calls port=5060 ;added bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ;

Greetings, I'm having a miserable time getting Asterisk working with FWD. All the samples show something like... exten => _7., .... How do I get Asterisk to wait until the user is finished dialing instead of trying as soon as it gets the second digit? I can use _7XXX, and dial the FWD 3-digit test numbers fine, but I'd like to be able to dial others... Same problem for outside

I'm sure there is no consistent way to reproduce this, but I'm hoping someone has some information. I have a time series we'll call y. The data gets updated every day, so I run a cron job that fits and predicts from an arima(0,0,1) X (1,1,1)_7 model. When I open R and run the script, it processes completely. If I call the script via a crontab entry R --no-save --slave <

Hi, Is it possible to use asterisk with Free World Dialup (FWD) ? Did someone manage to make it work? how? Best, -P -- __________________________________________________________ Sign-up for your own personalized E-mail at Mail.com http://www.mail.com/?sr=signup CareerBuilder.com has over 400,000 jobs. Be smarter about your job search http://corp.mail.com/careers

Hi all, On my Alpha box running Tru64 5.1, rsync-2.5.4 compiled by GNU gcc 3.0.1 anh GNU make 3.79.1 has this size: 4562848. The same version of rsync compiled by Compaq C compiler V6.3-028 and Compaq make has this size: 655424. It is about _7 times_ smaller! I did a quick test and they seem both working OK. Is that normal to see a huge difference in binary size produced by gcc and vendor's

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a "480 - user is not online" message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks

I'm hoping someone can help me out with this. I am basically just trying to separate my outbound and inbound calls into separate contexts instead of having everything in a single context. Any help would be appreciated. Perhaps I've missed something really obvious.... Here is the network layout: <remote> <--TDM400P--> <nattedbox> <--IAX--> <liveipbox>

Hopefully someone here can save my sanity. I have been trying to solve this problem for days now, but just cant put my finger on it. Im new to * so I have probably done something stupid! I have a TDM400P with one FXO module and a FXS module. The main problem I have is not being able to get the extension attached to the FXS module to ring or be able to make calls. It gets a dialtone fine but I

Does anyone know how to do this with the OH323 channel driver? I want the local (7 digit dialing) to go out an h323 that I have registered to a gatekeeper... can I do something like exten => _7.,2,Dial(OH323/ipofgatekeeper) -----Original Message----- From: Begumisa Gerald M [mailto:beg_g@eahd.or.ug] Sent: Thursday, September 09, 2004 2:37 PM To: Asterisk Users Mailing List - Non-Commercial

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

On 25/08/2019 15:50, Gregory P. Ennis wrote: > > On 25/08/2019 02:28, Gregory P. Ennis wrote: >> >> On Sat, Aug 24, 2019 at 3:21 PM Ian Mortimer <i.mortimer at uq.edu.au> wrote: >>> >>> On Sat, 24 Aug 2019, Gregory P. Ennis wrote: >>> >>>> On August 2, my desktop unit updated with >>>>

I was optimizing a 650MB using ferret (0.11.3) and I received the following error. I''ve seen some people have similar issues but I haven''t seen any resolutions. The contents of the index directory follow the error. Has anyone seen anything like this and found a resolution? Many thanks. /mnt/apps/search/releases/20070622175637/script/../config/../vendor/

To provide a bit more backwards-compatible (which is nice for eg. NFS- shared /home directories) try a few version-number based names. Eg., for "OpenSSH_7.3" the strings that are tried after "~/.ssh/config" are "_7.3", "_7", and "". --- ssh.c | 37 ++++++++++++++++++++++++++++++------- 1 file changed, 30 insertions(+), 7 deletions(-) diff --git