: 5301d4ba.3040...@laimbock.com
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On 17-02-14 09:26, Deka, Rajib IN MAA SL wrote:
Dear Forum,
I have encountered a similar issue as below in Asterisk 10.0.0.
Asterisk crashed while executing ?meetme kick all? CLI command from
manager
Dear Forum,
I have encountered a similar issue as below in Asterisk 10.0.0. Asterisk
crashed while executing meetme kick all CLI command from manager interface.
The link says the issue has been closed however I am not able to identify in
which release of asterisk this issue has been fixed.
Hi List,
We have a major issue while broadcasting DTMF using meetme application. We are
sending DTMF to asterisk using SIP INFO message with duration 160.
INFO sip:xxx@xxx SIP/2.0
Via: SIP/2.0/UDP xxx:5060
From: sip:xxx@xxx;tag=43
To: sip:xxx@xxx;tag=9753.0207
Call-ID: xxx@xxx
CSeq: 25634 INFO
Hello Forum,
Does any version of asterisk supports Malicious Communication Identification
(MCID) using IP standard 3GPP TS 24.616? If yes how can I enable or configure
it?
Regards
Rajib
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On Tue, Sep 3, 2013 at 4:17 AM, Deka, Rajib IN MAA SL
rajib.d...@siemens.com wrote
Hello List,
In our lab asterisk has crashed due to some unknown reason and it has been
restarted by safe_asterisk service. But before crash we can see lots of below
log entry (asterisk version 1.8.9.3).
Sep 3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error of
packet to
Hello List,
I tried to change the following parameters in sip.conf file, but looks like it
cannot be changed,
Defaut values:
;t1min=100
;timert1=500
;timerb=32000
I have changed to:
;t1min=100
timert1=100
timerb=6400
Sometime I can see too many retransmission of BYE to some of the UAs
Thanks a lot Dona and jg for your inputs.
I'll try to find some way to do this from Dialplan or AMI and let you guys know
soon. Please share if you have some more ideas.
Regards,
Rajib
Date: Tue, 11 Jun 2013 18:34:46 +0200
From: jg webaccou...@jgoettgens.de
Subject: Re: [asterisk-users]
Hello List,
I want to play an announcement for attended transfer calls. For example, A
calls B, B answers the call and transfers (attended) to C - once transfer
is complete B should hear an announcement saying you call has been
transferred. Is there any configuration in asterisk to implement
Hi,
We are facing an issue with asterisk in the case of call-Transfer scenarios.
Our requirement is to identify whether an incoming call is a fresh incoming
call or a Transferred call from some other clients.
We have a setup, where in the asterisk1.6 (as SIP server) is running in Linux
Hello List,
We are eagerly waiting for stable release of Asterisk 10 as it support most
awaited out of call messaging.
Can somebody please let me know when the stable release will be available for
download?
Regards
Rajib Deka
Siemens Ltd.
India
--
the tone and can avoid forwarding of
SIP INFO? I know I may be wrongly interpreted the things, Can somebody please
explain me the scenario, if possible?
Thanks
Rajib
From: Deka, Rajib IN MAA SL
Sent: Monday, August 29, 2011 3:34 PM
To: 'asterisk-users
Hello List,
We are using 'F' parameter in meetme Dialplan application to broadcast SIP INFO
(1 and 0) as DTMF tone to all the participants.
The DTMF configuration for all the connected SIP clients is SIP INFO.
The problem we are seeing, asterisk is taking some time to broadcast the SIP
INFO
Hello List,
Is it possible to send SIP:INFO to an active SIP channel using AMI action
PlayDTMF?
I tried to send a DTMF digit 1 to my SIP client, but I did not get any SIP:INFO
on wireshark trace.
In my sip.conf file dtmfmode=info.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran
Hi List,
Is it possible to configure an infinite ring timeout for queue in asterisk?
I mean, the caller should be able to be in queue until and unless he
disconnects the call.
Thanks,
Rajib
Important notice: This e-mail and any attachment there to contains
Hello all,
I found a problem regarding SIP presence in asterisk (1.6.2). The scenario is
not working properly for all users.
Our SIP client sends SIP:SUBSCRIBE to all the configured extensions in asterisk
during registration process. Asterisk replies with 200 OK for all SUBSCRIBE.
But if I run
: Deka, Rajib IN MAA SL
Sent: Tuesday, July 05, 2011 12:15 PM
To: 'asterisk-users@lists.digium.com'
Subject: SIP Presence not working
Hello all,
I found a problem regarding SIP presence in asterisk (1.6.2). The scenario is
not working properly for all users.
Our SIP client sends SIP:SUBSCRIBE
Hello List,
We are facing a problem in broadcasting DTMF from MeetMe.
Our client is sending DTMF with duration=160 (in SIP INFO) to asterisk but
asterisk is changing this header to different values like 162, 175 etc while
broadcasting to all the participants. Is it possible to restrict asterisk
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Deka, Rajib IN MAA
SL
*Sent:* Monday, June 13, 2011 6:44 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] asterisk queue 'ringall' stratagy
Hi List,
I have faced a problem in asterisk queue implementation.
I
Hi List,
I have faced a problem in asterisk queue implementation.
I configured a queue with 'ringall' strategy and 'ringinuse=yes' in
queues.conf. If three calls come to this queue in parallel, the logged in queue
agent used to get only one call (may be the first one), not all the calls
Hello List,
What version of DAHDi should be installed for CentOS Kernel version
2.16.18-194.el5.
We do not have access to yum in our network, so we need to install a specific
version with respect to kernel version.
Or, what update to be downloaded and applied to CentOS kernel to install a
Thank you very much Ruffell and Patrick.
The problem was basic. The OS was missing with correct kernel headers.
We installed correct kernel headers and its working fine now.
Regards,
Rajib
On Mon, May 30, 2011 at 02:29:37PM +0200, Patrick Lists wrote:
On 05/30/2011 10:03 AM, Deka, Rajib
Thank you John!
I too figure out the way using 'sip show subscription'.
Regards,
Rajib
Hello List,
Asterisk CLI command ?core show hints? gives the list of hint extension
configured and its presence status.
In command output there is a field called ?watchers? and it contains a
numeric
Hello List,
Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode
enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the
same.
Dialplan looks like
Exten = 100,1,MeetMe(100,dmF)
Sip.conf
dtmfmode=info
Regards,
Rajib
It's working now. I removed the 'm' flag from the meetme Dialplan. It was my
mistake. Asterisk is working fine.
Exten = 100,1,MeetMe(100,dF)
Regards,
Rajib
From: Deka, Rajib IN MAA SL
Sent: Wednesday, May 11, 2011 5:35 PM
To: 'asterisk-users@lists.digium.com
will useful to solve your requirement.
Regards
Dhaval
On Wed, May 4, 2011 at 1:13 PM, Deka, Rajib IN MAA SL
rajib.d...@siemens.com wrote:
Hello List,
We are running two asterisk machines in virtual IP as primary and secondary
server.
Initially virtual IP will be active in primary server
Hello List,
We are running two asterisk machines in virtual IP as primary and secondary
server.
Initially virtual IP will be active in primary server; during the failure of
primary secondary will get the virtual IP.
Is there any way to retrieve pending queue calls from primary to secondary, in
Hello List,
Please help with the following problem,
I have a situation, where I need to play an audio announcement to the caller
SIP channel once an attended transfer is successful. The attended transfer is
done from client. I can see a transfer event in AMI. I am not using 'T/t'
option in
,
Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote:
The requirement is little complicated as it is H/W specific.
Basically we are integrating a radio gateway (SIP) with asterisk. The
gateway will be
connected to a meetme room, so that any operator (with IP phone
registered as SIP user
more time than that to try out the suggestions!
On Wed, Apr 20, 2011 at 3:20 PM, Deka, Rajib IN MAA SL
rajib.d...@siemens.com wrote:
Thanks a lot Tony and Dhaval for your much appreciable suggestions.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor
Hello List,
I have seen from the following link that, for SIP channels there is no audio
communication possible in MeetMe with AGI_BACKGROUND.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
Currently we are using asterisk-1.6.2 and the problem still persists. Is there
any solution
Hello List,
Is it possible to run an AGI script in backgroung for all the associated SIP
channels in ConfBridge Application? If yes how?
This can be done using 'b' parameter in MeetMe for non SIP channels.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
...@inblrk77m1msx.in002.siemens.net,
Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote:
I have seen from the following link that, for SIP channels there is no audio
communication
possible in MeetMe with AGI_BACKGROUND.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
Currently we are using asterisk-1.6.2
; format=flowed
On 11-04-07 09:59 AM, Deka, Rajib IN MAA SL wrote:
Is the following trunk has development version of out-of-call messaging
capability, also what is the version of asterisk,
http://svn.asterisk.org/svn/asterisk/trunk/
I don't believe the branches has been merged into trunk, you
Hello List,
I have found that asterisk supports only forwards in-dialog MESSAGE method.
That is, if the MESSAGE method is sent within an active call.
But according our requirement we need to send MESSAGE method to the other leg
without being in a call (general stateless proxy forward). Is it
support (Deka, Rajib IN MAA SL)
2. Re: Iptables configuration to handle brute force
registrations? (Gilles)
3. Re: BRI Configuration help me (mahesh katta)
4. Re: Iptables configuration to handle brute, force
registrations? (Gilles)
5. Compiling asterisk using NDK build
, at 14:32, Deka, Rajib IN MAA SL wrote:
Is the following is the link for getting the source,
http://svn.asterisk.org/svn/asterisk/trunk/
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Hello List,
I have scenario as follows,
1. A call comes to queue.
2. Available agent will answer the call.
3. BridgeEvent wil be generated in AMI with channel1 and channel2.
4. Parse channel1 and channel two from the event and redirect them to a
meetme room,
Dialplan,
Exten =
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