Displaying 20 results from an estimated 4000 matches similar to: "No joy with my first AGI Python script"
2016 Aug 04
4
Removing mailbox and password prompt for voicemail
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
2014 Aug 07
2
Calls not hanging up
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
2016 Nov 22
3
Touch tone stutter
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
2015 Aug 25
4
Ringback issue
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
2016 Jul 30
3
Removing mailbox and password prompt for voicemail
If I remove the password, how can anyone access the mailbox if the
'mailbox' prompt is not played?
Nabeel
On 30 Jul 2016 3:19 p.m., "D'Arcy J.M. Cain" <darcy at vex.net> wrote:
> On Sat, 30 Jul 2016 06:43:47 +0100
> Nabeel <nabeelshikder at gmail.com> wrote:
> > I am using Asterisk voicemail on a CentOS 7 server. I would like to
> > be able to
2015 Jun 12
2
Voice mail and caller ID
I have this in my sip.conf:
exten => *98,1,Verbose(0,CALLERID number is "${CALLERID(num)}")
same => n,VoicemailMain(${CALLERID(num)}@LocalSets,s)
same => n,Hangup
However, my extensions are set up so that they always show the external
number, not the extension:
[foobar2](client-phone)
secret=xxxxxxxxxxxxxxxxxxxxxxxxxxxxx
callerid=Candace <5555551212>
2015 Feb 12
9
Is Asterisk a Linux only system?
I know that it runs on other systems but do other ports get the same
attention? I have been running it on a NetBSD server for about a year
now and while it mostly works it just crashes from time to time with no
explanation or core dump.
I have improved the situation by expanding my intrusion detection but
it still stops every few days or so. I have a cron job that tests for
it and restarts it
2015 Jun 18
1
setting outbound caller ID
Set(CALLERID(number)=XXXXXXXXXX) works here.
Also check with your VoIP provider what format they want for the number. (I
believe) most accept a 10-digit number, but I seem to remember reading
about the odd provider that wanted a leading "1".
On Thu, Jun 18, 2015 at 11:47 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote:
> On Thu, 18 Jun 2015 13:45:10 EDT
> kenner at
2015 Mar 12
2
Unstable phone connection
This is driving me to distraction. I have a switch with multiple
clients who are all working fine except for one and I can't figure out
what makes them different. I have tried every NAT setting in the ATA
(SPA112 ATA with 2 x FXS, 1 x LAN), stun server on and off, different
sip ports, different RTP ports and it still fails. I have left the
location with it working only to have it fail
2016 Sep 01
2
Multiple phones when one is unregistered
On Thu, 1 Sep 2016 11:02:57 +0200
Administrator TOOTAI <admin at tootai.net> wrote:
> > [Aug 31 21:52:00] WARNING[-1][C-0001fed5] pbx.c: No application
> > 'ExecIf' for extension (unauthenticated, 5555551111, 3)
> >
> > Is there a module that I need to load?
> >
> > In case it matters I am running Asterisk 11.23.0 on NetBSD 7.0.
>
>
2016 Aug 05
2
Toll free pattern matching
I have this in my config:
exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/tollfree/1${EXTEN})
exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/tollfree/${EXTEN})
exten => _NXXNXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN})
same => n,Dial(SIP/trunk/1${EXTEN})
exten =>
2016 Sep 01
2
Multiple phones when one is unregistered
On Tue, 30 Aug 2016 17:56:35 +0200
Administrator TOOTAI <admin at tootai.net> wrote:
> Something like
>
> exten => 5555551111,1,Verbose(Door buzzer calling)
> same => n,Set(toRing=)
> same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
> USE"]?Set(toRing=${toRing}&SIP/user1)
Failed. I checked the online docs and the syntax seems to
2015 Aug 11
3
One way audio - doesn't seem to be NAT issue
I have been banging my head against the wall for weeks now on this
one. I have a switch running NetBSD and Asterisk 11.19.0 although I
have had this problem on older versions as well. I, and my users, can
call out, we can receive calls, quality is excellent but I cannot talk
with one user. The different elements are as follows:
The switch as described above which is in a server room on the
2016 Aug 30
12
Multiple phones when one is unregistered
I have an extension that looks like this:
exten => 5555551111,1,Verbose(Door buzzer calling)
same => n,Dial(SIP/user1&SIP/user2&SIP/user3)
The idea is that any of the three users can answer the phone to let
someone in. The problem is that if, say, user2 unplugs his phone then
the call immediately goes to his voice mail and the other two do not
have the ability to open the door.
2017 Apr 20
2
Voicemail asking for login
On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote:
> This is just screaming "configuration mismatch" -- or, possibly, "latent bug
> whereby things parsed in separate places should be treated the same, but are
> actually getting treated differently".
I really don't want to be the "my system isn't working so there must be
a bug in Asterisk" guy
2015 Aug 15
2
One way audio - doesn't seem to be NAT issue - SOLVED!
On Sat, 15 Aug 2015 16:30:39 +0800
Michael Dupree <michael at easybitllc.com> wrote:
> Not 100% ure, but maybe play with the canreinvite or directmedia
> settings.
Yes! That was it. Just for future searches here is what I did. I
added "directmedia = no" in sip.conf. This fixed the issue.
I believe that Asterisk was getting confused when one leg was inside
NAT and the
2016 Aug 30
2
Multiple phones when one is unregistered
On Tue, 30 Aug 2016 10:39:14 -0400
Eric Wieling <ewieling at nyigc.com> wrote:
> The dialplan below cannot go to voicemail, either something else is
Of course not. It's the individual extensions that have voice mail. I
have a similar problem when one of those destinations is a cell phone
but I know that there is no Asterisk solution for that problem. If the
cell phone answers and
2016 Nov 23
2
Touch tone stutter
On 2016-11-22 07:49 PM, Pete Mundy wrote:
>
> One direction that may be worth exploring further is his ATA's config (or perhaps swapping it for a different model). Eg adjusting echo cancellation or line impedance settings.
I have to be careful here as I auto-provison these devices and changes
would propogate to every user. Echo cancellation is off. Do you think
it should be on?
2016 Aug 04
4
Removing mailbox and password prompt for voicemail
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
2018 Jun 08
3
T-38 re-invite issue
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