similar to: asterisk-users Digest, Vol 81, Issue 21

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 Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery

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.

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 waiting in the queue at a time. Once the agent answers the call the next call is displayed. I

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>

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 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

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 available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

Hello List, I have a requirement like, Only one UA can register at a time (the registration should be independent of IP). If some other UA tries to register from a different 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

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 dial() command. The transfer is completely on client side using SIP signaling. 1. A calls B 2. B

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 specific version of DAHDi. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor,

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 possible to do this in asterisk using some tricks? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran

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. Please help. https://issues.asterisk.org/jira/browse/ASTERISK-15741 With best regards, Rajib

Hello List, I am facing some issue while passing DTMF (RFC2833 set globally in sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two users tries to pass DTMF simultaneously at the same time from their phones only one DTMF is detected in asterisk and broadcasted to other users. Other DTMF lost somewhere. We have tested only with sip phones. Can someone help me with this, or

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

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

Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote: > Hi > > We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for > queues. > > For a particular customer, when I run queue show <queue_name> I get the > following numbers: > > <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s

On Tue, Jan 27, 2015 at 4:14 PM, symack <symack at gmail.com> wrote: > Hello Everyone, > > I am required to write a java program that will get our asterisk to: > > * Query the database for phone numbers > * Loop through numbers and dial > * Play message > * Get dial pressed response > - If 1 = Yes > - If 2 = No > - If 3 = Connect to Agent

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod at mogl.com> wrote: > For those that were interested I have attached the kamailio.cfg which we > have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the > following yum packages: > > kamailio.x86_64 4.2.1-4.1 > @home_kamailio_v4.2.x-rpms > kamailio-auth-ephemeral.x86_64

On Thu, Mar 5, 2015 at 2:29 AM, Dmitry Melekhov <dm at belkam.com> wrote: > Hello! > > Just installed asterisk 13.2.0 and see many such messages in log, I see them > in console during calls, really something like this: > > > -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", > "SIP/6166 at asterisk") in new stack > == Using SIP