Displaying 20 results from an estimated 10000 matches similar to: "Failed to authenticate device - who?"
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
2015 Mar 12
0
Unstable phone connection
D'Arcy J.M. Cain
If the device is registering and then dropping there are several usual
items.
The router may be closing the ports on the device.
The router may have a AGL SIP helper that is causing issues.
Make sure that the device is sending out keep alive packets.
Shut down any AGL helpers on the router.
Make sure that the site is not double NATing
Try using a stun
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
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
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
2015 Jun 12
0
Voice mail and caller ID
Try this for CHAN_SIP:
same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox
same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a
mailbox defined log into it
If you are using PJSIP it's more complex
same => n,Set(EndPoint=${CHANNEL(endpoint)}) ; Get the peer
same =>
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
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
2019 Nov 27
2
Faxes stopped working - AMI issue?
I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is
working fine with a few minor tweaks except outgoinf fax. Incoming
works fine.
I do outgoing faxing through an AMI call. Here is the output from the
security log:
[Nov 27 06:16:05] SECURITY[101222] res_security_log.c:
2015 Jun 16
0
howto copy a voicemail message to another machine ?
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 when I learned
Asterisk. It was dead simple. Rather than creating some sort of Rube
2015 Feb 12
0
Is Asterisk a Linux only system?
I would love to run Asterisk on a BSD system. I do not know of any developers actively working on Asterisk on a BSD platform, though my knowledge isn't comprehensive.
It may be worth talking to the people doing the packaging for various BSD platforms, to see how involved they are, or if they know of people developing it directly. jnemeth at netbsd for pkgsrc, madpilot at freebsd for
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
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.
>
>
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 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 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 =>
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
2017 Apr 19
2
Voicemail asking for login
On 2017-04-18 08:17 PM, Pete Mundy wrote:
>> On 19/04/2017, at 7:58 am, D'Arcy Cain <darcy at VybeNetworks.com
>> <mailto:darcy at VybeNetworks.com>> wrote:
>>
>> <snip>
>> Everything looks the same as another one that works except for two
>> things. The one that works doesn't have the "Probation passed" lines.
>> I am
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
2015 Jun 12
1
Voice mail and caller ID
On Fri, 12 Jun 2015 11:49:05 -0700
John Kiniston <johnkiniston at gmail.com> wrote:
> Try this for CHAN_SIP:
>
> same => n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
> same => n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the
> mailbox same => n,VoicemailMain(${MailBox}@LocalSets,s) ; If we
> have a mailbox defined log into it
Perfect.