search for: deka

...only one call (may be the first one), not all the calls waiting in the queue at a time. Once the agent answers the call the next call is displayed. I want to display all the waiting calls on the agent's desktop. Is it possible to do, if yes how? Please help me with this. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

...risk 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 Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com&...

...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 available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe. Kind Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

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, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

...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 possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

...ured and its presence status. In command output there is a field called "watchers" and it contains a numeric value of number of subscriptions' registered for that particular extension. So, is there any CLI command to check who the watchers for an extension are? Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

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 [2002:c117:a683::c117:a683]:20940: Address family not supported by protocol chan_sip.c: Purely numeric

...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 specific version of DAHDi. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

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 case primary fails? Does asterisk provide any interface to do it or we have to write some application

Thank you Paul. I have downloaded the code. How out-of-call messaging can be configured in the Dialplan? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Thu, 07 Apr 2011 10:14:37 -0400 From: Paul Belanger <pabelanger at digium.com> Sub...

.... A calls C 4. B is in hold 5. C answers 6. B transfer A->C 7. Play an announcement to A 'your call transferred' 8. A and C bridged. I have enabled "xfersound = <some audio file>" in features.conf. But it's not working once I transfer the call. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

...ent IP using the same credentials, it should be blocked by asterisk. We do not want to permit or block any IP or subnet in sip.conf. Following is an example of sip user configuration, [217] type=friend username=217 host=dynamic context=outgoing allowsubscribe=yes qualify=yes Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com<http://www.siemens.com> Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com ________________________________ Important notice: This e-mail...

...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. Please help. https://issues.asterisk.org/jira/browse/ASTERISK-15741 With best regards, Rajib Deka Siemens Ltd. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140217/adb2b76d/attachment.html>

...any configuration option that can resolve this problem? I want asterisk receive the DTMFs send at the same time and to pass those either by queuing them or by some other means. We can not use confbridge at this moment as we have developed the application on meetme. Please help! Regards, -- Rajib Deka Sr. Programmer Siemens Ltd. Chennai, India -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131116/fa201913/attachment.html>

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 from changing this header value or this is a common behavior of all the PBXs. Regards, Rajib

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 this behavior? I have not

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 <webaccounts at jgoettgens.de> Subject: Re: [asterisk-users] announcement to be played for attended transfer call To: Asterisk Users Mailing List -

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 if UA is unreachable. Is there a way that I can reduce the number of retransmission of BYE message?

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131111/5bf768a4/attachment.html>