similar to: howto store local exchange prefixes ?

My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message. I'm thinking of: same =>n,VoiceMail(vm,u) same =>n,System(ssh myserver "emailVM

On 06/16/2015 11:52 AM, D'Arcy J.M. Cain wrote: > On Tue, 16 Jun 2015 11:35:26 -0400 > sean darcy <seandarcy2 at gmail.com> wrote: >> My asterisk server is in the cloud. Figuring out how to send an email >> is too much brain damage. So i can't use the email feature that's >> built into voicemail. > > Really? That was one of the first things I did

I''m having a problem switching from cookie sessions (the default) to active record sessions. I''ve created the sessions table via the rake task, uncommented the line config.action_controller.session_store = :active_record_store in my environment.rb file, cleared out my browsers cookies, restarted the server (natch), but still the it''s using cookie_store instead of

1.6.1 svn 190575: CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" menuselect make[1]: Entering directory `/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect' gcc -m64 -march=native -mtune=native -floop-interchange -floop-strip-mine -floop-block -c -o

Patches 1 & 2 were posted previously here: https://www.redhat.com/archives/libguestfs/2016-May/msg00134.html Patch 3 is a hack so I can test this using my own version of qemu (the `-L ?' stuff is not upstream). Patch 4 is where the real action takes place: Replace the caching of qemu features in blob-like files with a sqlite database. Probably the best way to approach this patch is to

With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. extensions.conf: [meetme] exten => 2663,1,MeetMe(,De) exten => 2663,n,Hangup() exten => 2666,1,MeetMe() exten => 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set a PIN. Hangup. Then users would dial

On Thu, 4 Aug 2016 14:03:39 +0100 Nabeel <nabeelshikder at gmail.com> wrote: > I should add, a password is *always* asked if a password has been set. > There isn't a way to bypass that. Then something is wrong. http://darcy.vex.net/star98.mp3 -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net VoIP: sip:darcy at Vex.Net

On 2017-04-19 11:57 AM, J Montoya or A J Stiles wrote: > I fished this out of an old extensions.conf from a defunct project. It might > be relevant to your use case: > > exten => 1571,1,NoOp(Call to 1571: voicemail retrieval) > exten => 1571,n,AGI(lookup_caller_id.agi,${CALLERID(num)}) > exten => 1571,n,NoOp(CLID is ${clid}) > exten =>

I am hoping someone else has seen this and can offer a solution or at least a direction to investigate. I am running 11.23. Most of my clients are fine but one has a strange behaviour. He has a Grandstream HT701 like most of my clients who use an ATA. He can make call and they are crystal clear. However, when he tries to use phone menus ("dial 234 for John Doe" for example) it

Hello, I'm trying to figure out how to keep a mirror and 7 day incremental backup between to 2 mount points. I want to rsync everything from /mnt/production/ to /mnt/backup/production/ on the same server. Nothing fancy. But , I Want to utilize the --delete option for files that no longer exist on /mnt/production/ that have been on /mnt/backup/production/ for longer then 7 days. I

On 4 August 2016 at 13:18, D'Arcy J.M. Cain <darcy at vex.net> wrote: > > Let's get this straight. You call yourself from any phone in the world > and press '*' while listening to the message, you wind up in your own > mailbox and you believe that means that you don't need a password? Do > you think that the phone system somehow knows that it is you

>>>>> D'Arcy Cain <darcy at VybeNetworks.com> writes: >> Ie after both sides select t38, until they agree on the t38 terms. > OK, so does that mean that setting it to 25000 should leave time for the > re-invite or does the timeout start after that. As I wrote above, after that. After the sip/sdp. -JimC -- James Cloos <cloos at jhcloos.com>

This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. # asterisk -x "core show channels" Channel Location State Application(Data) SIP/thinktel-0000000 (None) Up AppDial((Outgoing Line)) SIP/4164251212-00000 4165555555 at LocalSets Up Dial(SIP/thinktel/4165559999) 2 active

My last problem was nicely solved through this mailing list so hopefully this new problem will have the same happy outcome. My situation is that I have many extensions. Here is a sample: [client-phone](!) type=friend host=dynamic secret=XXXXXXXXXX dtmfmode=auto disallow=all allow=ulaw allow=gsm allow=g723 allow=ilbc subscribemwi=no [4165555555](client-phone) secret=xxxxxxxxxxxxxxxxxxxxxxxxxx

It's very simple but it doesn't work. Here's the entire script. #! /usr/bin/python import sys env = {} def comm(cmd): sys.stdout.write(cmd.strip() + '\n') sys.stdout.flush() return sys.stdin.readline().strip() while 1: line = sys.stdin.readline().strip() if line == '': break key,data = line.split(':') if key[:4] == 'agi_':

> CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan.

Dear R users, I am trying to write myself a loop in order to produce a set of 20 length frequency plots each pertaining to a factor level. I would like each of these plots to be available on the same figure, so I have used par(mfrow = c(4, 5)). However, when I run my loop below, it produces 20 plots for each factor level and only displays the last factor levels LF plots. I'm fairly new to

On 2017-04-18 02:42 AM, Pete Mundy wrote: > Try this: > > asterisk -r > core set verbose 10 > [get user to trigger fault] > [examine console output, and post to list if still unclear] > > If you don't solve it yourself, then we'll be able to help further once > we've seen the output. I can't see much more than at my previous debug level but here it is

Hi, I''m using an sqlite3 database with Rails and have some trouble running unit tests What I''ve done is * rails my_app * edited database.yml to use sqlite3 adapter, db/my_app_development and db/my_app_test * ruby script\generate model user * edited model class User < ActiveRecord::Base validates_presence_of :name validates_uniqueness_of :name end *

I have an error sending to a specific fax number. It may be more than one but this is the one I investigated. It seems the delay for the SIP negotiation in T.38 was initiated after 6 seconds, however, our system sent the BYE after only 4 seconds, possibly cutting the call before all the communication necessary for the negotiation was completed. Here is the trace from our provider showing their