Hi,
I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the
user and call AgentLogin. But after that when I call AgentRequest I keep
getting Agent '1234' is busy.
If I put a delay of 5 second or more before calling AgentRequest then it
works most of the times. Here's my dialplan:
Hi All,
I have been working on a project where I need to record a call in Asterisk
and then split the recording into multiple audio files based on a presence
of particular sound (i.e. beep) in a recording.
I know this is out of scope for Asterisk but I wanted to benefit from
someone else's
Hello,
Thank You Paul for your reply,
The registrations in my setup are not duplicated, the 'secret' field in the
realtime table is empty, which causes Asterisk to not authenticate requests
from my Kamailio. Kamailio handles registrations, and also routes the
traffic to Asterisk using
The output of a failed incoming fax:
http://pastebin.com/S58j0WbW
A failed outgoing fax:
http://pastebin.com/eqveVZgK
It seems to me that I installed it.
fax show licenses
Fax Licensing Information
==
Free fax licenses: 1
Total licensed
Hello Everyone,
Today we observed asterisk sending two invites for the initial call before
the call was established (ie, not re-invites). There were no changes made
to the configuration for a very long time, and was kind of confused when
seeing this action. Can someone please suggest where to
Okay, tried reverting to Asterisk 11.10.2. I didn't change the realtime
table yet, but now when calling from websocket client to another websocket
client, cli says:
WARNING[30620][C-]: chan_sip.c:11056 process_sdp_a_dtls:
Unsupported fingerprint hash type 'sha-2' received on dialog
Hello,
How can I read RTP ports from CLI (to double check what could be
included in rtp.conf file) ?
sip show settings do not provide the answer.
Regards
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On Tue, Aug 12, 2014 at 4:17 AM, Olli Heiskanen
ohjelmistoarkkite...@gmail.com wrote:
Hello,
Thank You Paul for your reply,
The registrations in my setup are not duplicated, the 'secret' field in the
realtime table is empty, which causes Asterisk to not authenticate requests
from my
Hello,
I have my provider dropping the calls after 41 seconds of not receiving any
RTP from my asterisk. Obviously there is no RTP back when the caller is
leaving a message in the voicemail. Is it possible to have asterisk
generate some RTP packet back?
Leandro
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On Tue, Aug 12, 2014 at 1:33 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:
Hi,
I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the
user and call AgentLogin. But after that when I call AgentRequest I keep
getting Agent '1234' is busy.
If I put a delay of 5 second or
On Tue, Aug 12, 2014 at 10:57 AM, Richard Mudgett rmudg...@digium.com wrote:
On Tue, Aug 12, 2014 at 1:33 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:
Hi,
I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the
user and call AgentLogin. But after that when I call
Hey there
i'm trying to get an Asterisk 11.11 with encryption working with my
Grandstream phones. But i stuck.
To avoid NAT problems i'm using IPv6
Just with TCP/TLS it's working fine. Only the SRTP funktion is not working.
Asterisk tells me
WARNING[6938]: chan_sip.c:3906 __sip_xmit: sip_xmit
Thank you for the response Richard and Matthew! It's good to hear that you
are working on fixing the 5s delay. I was really puzzled by it and found
the idle time by trial and error. Is there any documentation of these new
behaviors?
Coming back to the main issue, I am getting agent busy when the
On Tue, Aug 12, 2014 at 11:24 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:
Thank you for the response Richard and Matthew! It's good to hear that you
are working on fixing the 5s delay. I was really puzzled by it and found
the idle time by trial and error. Is there any documentation of
Hello,
A couple of questions in relation with Asterisk 12 on Debian Wheezy.
1. Can paquet libpjproject-dev (from wheezy-backport) be installed as
the sole binary to add PJSIP stack to Asterisk 12 (compiled from
source) ?
2. When compiling PJPROJECT from source (see
On Tuesday 12 Aug 2014, Olivier wrote:
Hello,
A couple of questions in relation with Asterisk 12 on Debian Wheezy.
1. Can paquet libpjproject-dev (from wheezy-backport) be installed as
the sole binary to add PJSIP stack to Asterisk 12 (compiled from
source) ?
2. When compiling
Hello. I tryto use Statis at my dialplan to run my app (a)
When Statis is running from making call ( I dial from softphone some exten
and run dialplan context where call Statis(MyApp)) Asterisk responsed:
ERROR[61517][C-0019]: res_stasis.c:852 stasis_app_exec: Stasis app
'MyApp' not
Yuriy Gorlichenko wrote:
Hello. I tryto use Statis at my dialplan to run my app (a)
When Statis is running from making call ( I dial from softphone some
exten and run dialplan context where call Statis(MyApp)) Asterisk
responsed:
ERROR[61517][C-0019]: res_stasis.c:852 stasis_app_exec:
There is right at 500 ms between the two invites. You are seeing a
retransmission due to a lack of response to the first INVITE in time. This
is normal, correct, and expected behavior. The retransmission can occur
even sooner in the case where QUALIFY is used to determine that the
endpoint
On Tue, Aug 12, 2014 at 12:38 PM, Olivier oza.4...@gmail.com wrote:
Hello,
A couple of questions in relation with Asterisk 12 on Debian Wheezy.
1. Can paquet libpjproject-dev (from wheezy-backport) be installed as
the sole binary to add PJSIP stack to Asterisk 12 (compiled from
source) ?
Thanks Scott, Restarted all the machines since there uptime was 8 years :).
Everything works ok now.
Kind Regards,
Nick.
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New to Asterisk? Join us for a
Eric is correct. There is no way to send dtmf while the call has not been
answered.
But us very confusing the read command, in specific option = n(noanswer)
to read digits even if the line is not up
My AGI line is the following
$AGI-exec(READ,umenu,VARXX,1,n,2,7);
The command works, but there
I do not know, maybe some of the other channel drivers sccp or sip support it.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael Visser
Sent: Tuesday, August 12, 2014 7:24 PM
To: Asterisk Users Mailing List - Non-Commercial
I am talking about sip on asterisk 11.10.2
rv
2014-08-12 19:28 GMT-04:00 Eric Wieling ewiel...@nyigc.com:
I do not know, maybe some of the other channel drivers sccp or sip support
it.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On
Le 12 août 2014 18:49, A J Stiles asterisk_l...@earthshod.co.uk a écrit
:
On Tuesday 12 Aug 2014, Olivier wrote:
Hello,
A couple of questions in relation with Asterisk 12 on Debian Wheezy.
1. Can paquet libpjproject-dev (from wheezy-backport) be installed as
the sole binary to add
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