ers@lists.digium.com>
Subject: Re: [asterisk-users] Pass CallerId/Privacy info from A Leg to B Leg
On Thu, Aug 17, 2017 at 07:28:00AM +, Grant Bagdasarian wrote:
> Is there an option to give to the Dial command, or another variable to set,
> to make Asterisk copy such information to the B Le
Hi,
I'm using Asterisk to bridge the incoming call to another destination using the
Dial command.
However, when an anonymous call comes in then privacy information is not passed
into the B Leg.
For instance, the Privacy header and P-Asserted-Identity aren't copied to the B
Leg.
Is there an
Hello,
Does the asterisk-java library (https://github.com/asterisk-java/asterisk-java)
work with the latest LTS version of Asterisk?
I couldn't find information about the supported asterisk versions.
We're currently using the asterisk-java.1.0.0.m3 version on asterisk 1.6 and
are planning to
Hello,
We have a few Asterisk 1.8.14.1 boxes which occasionally suffer from audio
being lost in a bridged call.
So, the inbound and outbound channels are talking to each other for a few
minutes, no problems so far, and then suddenly they can't hear each other
anymore.
These calls are recorded
Hello,
Is it possible to log the raw signaling of Dahdi channels to a log file?
--
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Hello,
We're running Asterisk 1.8.14.1 and our carrier requires us to send a 200 OK
for OPTIONS request in order for them to keep sending traffic to our endpoints.
Asterisk is currently replying with 404 messages, and their SBC only accepts
200 OK responses.
How do I configure asterisk to
] Asterisk LTS segment faults
On Wed, Oct 8, 2014 at 9:35 AM, Grant Bagdasarian
g...@cm.nlmailto:g...@cm.nl wrote:
Hello,
Does anyone know how frequent segment faults occur in the current LTS release
(version 11) and in the future LTS release (version 13)?
We are currently using 1.6, which
Hello,
Does anyone know how frequent segment faults occur in the current LTS release
(version 11) and in the future LTS release (version 13)?
We are currently using 1.6, which frequently throws unexplained segment faults,
that's why we are considering to upgrade to the latest LTS version.
--
a knowledge about his hardware (microphone,
speaker, distance etc.).
---
Dennis Guse
On Tue, Aug 26, 2014 at 1:30 PM, Emiliano Vazquez
emilianovazq...@gmail.commailto:emilianovazq...@gmail.com wrote:
El 26/08/14 a las 05:33, Grant Bagdasarian escibió:
I’m new to Echo Cancellation and I
Hello,
Could someone explain to me what this means?
asterisk[30269]: segfault at 0008 rip 2aaac8b388f2 rsp
40a75910 error 4
Also, would this segfault crash the whole Asterisk process or will Asterisk
continue to run?
Is it possible this would affect/disconnect SOME
of unexplained segfaults in
1.6.X, haven't seen any in 1.8.22.0 (I'm afraid to upgrade since I finally
found a stable version)
You should, also, have you heard of FreeSWITCH? IMO much more stable PBX
software.
Thanks
On Thu, Aug 28, 2014 at 5:45 AM, Grant Bagdasarian
g...@cm.nlmailto:g...@cm.nl
Hello,
I'm new to Echo Cancellation and I was wondering how it is handled/works on
pure VoIP networks using Asterisk?
I did some research on the internet about EC on VoIP networks, but I can't
really put a grasp on it.
We currently have some Echo Cancellation chips on our Digium cards, but are
Hello,
Using Ubuntu Server 12.04 and Asterisk 11.2.1.
I'm getting the following error when trying to start asterisk:
(Syslog) kernel: [ 1032.713864] asterisk[26918] trap invalid opcode
ip:7fc272923076 sp:7fff928cf1b0 error:0 in codec_ilbc.so[7fc272921000+e000]
We were running Asterisk on a
Hello,
We have a Kamailio SIP Proxy in front of our Asterisk cluster for incoming
calls from our carrier.
The sip.conf looks like this:
[kamailio1]
type=friend
host=10.0.0.1
context=incoming
disallow=all
allow=alaw
All calls hit the incoming extension. In the extensions.conf we have multiple
10:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dedicated hangup extension h
On 28 Aug 2013, at 09:50, Grant Bagdasarian g...@cm.nl wrote:
Hi Grant!
I do not know of a way to have multiple 'h' extensions in the same context.
But you can easily
...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: Wednesday, August 28, 2013 3:51 AM
To: asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com
Subject: [asterisk-users] Dedicated hangup extension h
Hello
Hello,
I'd like to use the AMI interface to originate a call to a context in a
dialplan, and handoff the dial control to the context.
Whenever I execute the below action, the recipient does ring, but when I answer
it dials the recipient again. I believe this is because once answered the
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI
Originate
Looks correct to me
2013/6/19 Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl
Hello,
I’d like to use the AMI interface to originate a call to a context
the Dial
application has reached its timeout.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: Wednesday, June 19, 2013 11:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Handoff dial control to dialplan after AMI
Originate
On Wed, Jun 19, 2013 at 4:00 PM, Grant Bagdasarian
g...@cm.nlmailto:g...@cm.nl wrote:
Why can't I execute any more dialplan after the Dial application? The scenario
Hello,
I'm trying to execute a stored procedure on a MSSQL Server from the dial plan,
but it's not working. I'm getting the following error: Unable to execute
query
Asterisk has been compiled with UnixODBC, and I've done the necessary
configurations in func_odbc, res_odbc and odbc.ini.
Note, that writing CDRs using ODBC to a MSSQL database does work. So I don't
know why this doesn't.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: Friday, June 14, 2013 2:43 PM
To: asterisk-users
Hello,
I'm researching the possibilities of multiple communication platforms like
Asterisk and FreeSwitch for handling a dynamic sequence of applications to
execute, like Playback, Read, etc.
This only applies to originating a call from an external application by using
the AMI Manager and the
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Executing a dynamic sequence of applications
On 05/30/2013 04:46 AM, Grant Bagdasarian wrote:
Hello,
I'm researching the possibilities of multiple communication platforms
like Asterisk and FreeSwitch for handling a dynamic
Hello,
I'm trying to change the voice during a spoken text:
exten = _X.,1,Answer
exten = _X.,n,MRCPSynth(Hello, my name is Daniel. I have a Dutch companion.
###\voice=Xander\ Hallo, mijn naam is Xander.,p=defaultl=en-GB)
exten = _X.,n,Verbose(1, ${SYNTHSTATUS})
exten = _X.,n,Hangup
This exact
, 2013 10:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How does Asterisk handle ACK's?
12 mar 2013 kl. 16:54 skrev Grant Bagdasarian g...@cm.nlmailto:g...@cm.nl:
Hello,
I'm noticing strange behavior in one of our Asterisk nodes where the ACK
The problem seems to have been fixed with version 11.2.1. The ACK is now
correctly sent to the address in the Contact header of the 200 OK.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: Wednesday, March 13
Hello,
I'm noticing strange behavior in one of our Asterisk nodes where the ACK is
always sent to the proxy, but RR is not enabled for calls.
The proxy drops the ACK.
I'm using the AMI interface to originate a call:
Action: login
Username: myusername
Secret: mypassword
Events: on
Action:
Hello,
Our supplier requires the From header of a SIP INVITE to contain certain data
so the call is placed with a private caller id.
It needs to be like this: From:
sip:anonymous@anonymous.invalid;user=phone;tag=123455667
How do I configure Asterisk to dial anonymously?
Regards,
Grant
--
Hello,
I quite don't understand how to send a recorded message during a call off to an
HTTP handler using HTTP POST.
How do I access this file/audiostream in the dialplan?
I tried this:
exten = rpm,1,Set(RecordedPersonalMessage=${EPOCH})
exten =
/soundfragmenthandler.ashx)
Not sure if it's the best way to do, but it works.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant Bagdasarian
Sent: dinsdag 15 januari 2013 12:01
To: asterisk-users@lists.digium.com
Subject: [asterisk-users
Hello everyone.
The share is working and I'm now able to play audio files from a windows share.
Thanks everyone for the help!
--
_
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New to Asterisk? Join us
the whole path of the recorded file and
use it as a parameter in the CURL application?
About the streaming, no I haven't figured it out yet. I'll take a look at
app_ices. I hope it's not deprecated. Thanks!
On Wed, Jan 9, 2013 at 2:54 AM, Grant Bagdasarian GB at
cm.nlhttp://lists.digium.com
Regarding the streaming of audio.
I thought of another approach, but I'm not sure if Asterisk will allow it.
When playing a file they're read from /var/lib/asterisk/sounds/en/.
Is it possible to change this directory to a network directory hosted on a
windows environment?
--
Hello Users,
I've been searching for a couple of hours now but I can't find the answers to
my questions, so here they go:
1) Is it possible to stream audio files from a webserver during a call by
configuring this in the dialplan? Something like
Hello,
For some reason I did not receive any replies related to my question by mail,
but I found the topic back on the online mailing archives. I hope by supplying
the same subject this email will be logged in my previously created topic
instead of a new one. If it does not, I apologize.
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