similar to: [Asterisk-Dev] CVS HEAD behavior change: Beware!

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the

Hi, although setting up voicemail for busy and unavailable should be easy, things aren't working the way they should in my configuration (asterisk 1.2.14 bristuffed): Here's the relevant part of the extensions.conf: exten => 56830976,1,Answer() exten => 56830976,2,Dial(SIP/hbaumgart,20,tr) exten => 56830976,3,VoiceMail,u76 exten => 56830976,4,Hangup exten =>

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

Tie line is type em_w (old school stuff) to TE110P. Phone rings on the asterisk side, but the calling party does not hear the ring through sound. If I pick it up within the first two rings it goes through and I can talk otherwise our old switch drops the call. Anyhow...here is my config if anyone can shed some light on it. It used to work with HEAD a few weeks ago. -Matt

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

The patches from Mantis issue #752 (extended screening/privacy support for Dial) have been merged into CVS HEAD. As part of this patch, the 'n' flag to Dial got changed to be used as part of the privacy features, instead of being the 'dont jump to +101' flag. That flag is now 'j'. Sorry for the inconvenience, but sometimes things have to change in the development

I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing. Below is a dialplan snippet and the resulting CLI output. This is running in an 'h' extension. Noop(DIALSTATUS=${DIALSTATUS}) Noop(CDR(disposition)=${CDR(disposition)}) -- Executing [h at pbxmax-dial-simple:1] NoOp("SIP/msx_01-0000005b", "DIALSTATUS=ANSWER") in new stack

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

It really is unclear what is claimed to be a bug here. But see https://stat.ethz.ch/pipermail/r-devel/2007-May/045592.html for why the bug is not in R: your old and new data do not match. Your fit is to a category. [The problem with the web interface to R-bugs was reported last week: it is being worked on.] On Mon, 30 Apr 2007, r.darnell at uq.edu.au wrote: > This is a multi-part

Hi All, Asterisk 1.4.12 on CentOS 5 Sorry for a question that I'm guessing is obvious to most of you. I'm trying to revamp my dialplan. When I first created it, I had something like: exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => s,2,Dial(${rgMain},${RINGTIME},t) exten => s,3,VoiceMail(main at default) exten => s,103,VoiceMail(main at default)

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hi Folks, I have posted a couple of message to the list and do see them, even after waitin for long time (2 days). Can someone please point me to the rules for posting to this list? I think it had to do with the subject that I was looking for. I was looking for IAX terminiation service that can handle high volumes. Thanks John. --------------------------------- Yahoo!

hi listmembers, please test my new patch to chan_sip.c which is to make call pickup on the snom phones (and maybe other phones that support 'INVITE/Replaces') work and make comments in the bugtracker http://bugs.digium.com/view.php?id=3644 so it can make its way into the cvs. this patch sipsubscribe-20050715.rev779.txt enables: * monitoring of other lines (using the 'hint'

On Wed, 05 Jan 2005, CVS User ttroxell wrote: > @@ -70,6 +70,10 @@ > chown logcheck /var/lock/logcheck > /dev/null 2>&1 || true > fi > > + # fix for #284788 > + # update timestamp on cron > + touch /etc/cron.d/logcheck || true > + > ;; > > abort-upgrade|abort-remove|abort-deconfigure) on a box with a broken coreutils install

Hi guys, What is different and the "context" to play unavail message and busy message? When a SIP connection is unregistered, voicemail will play unavail message, right? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040913/1a2d1c81/attachment.htm

Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. #!/usr/bin/perl use strict; use

Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto("SIP/111-086497c8", "SIP/113-08674628|dynamic-nway|111|1") in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting

Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and 1.4.7.1 on FreeBSD 6.2) [general] priorityjumping=yes With n+101: exten => 1337,1,Dial(SIP/zytek,5,Ttj) exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS}) exten => 1337,n,Hangup -- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack -- Called zytek

I am trying to construct a macro for long distance dialling. I have two internet feeds, I have all routes including Teliax on Internet A and a static route to Voxee on Internet B. I thought I could use the dialplan entry below which uses the ChanIsAvail() command to check the connection, but this returns the provider but not the username, so I don't understand how to use this for real

I have the following in my dialplan, counts the number of loops and when it hits greater then 5, exit. It works, but errors initially with, "syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_LP or tolken; Input: +1". Could somebody tell me why? Thanks: ; **************************************** ; Setup a varriable to count the number of ; times the message has been