similar to: Asterisk changing SIP INFO dtmf duration

Displaying 20 results from an estimated 11000 matches similar to: "Asterisk changing SIP INFO dtmf duration"

2011 May 11
2
no audio with SIP:INFO in meetme
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
2011 Apr 20
2
No voice in MeetMe for SIP with
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>
2011 Apr 19
3
No voice in MeetMe for SIP with AGI_BACKGROUND
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.
2011 Jun 13
3
asterisk queue 'ringall' stratagy
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
2011 Apr 19
1
ConfBridge and AGI
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.
2011 May 30
2
DAHDi installation problem
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,
2011 May 25
2
asterisk hint SIP presence
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
2011 Apr 08
0
asterisk-users Digest, Vol 81, Issue 21
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
2013 Sep 03
3
Asterisk crash
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
2011 May 04
2
asterisk HA for queue calls
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
2011 Apr 07
4
asterisk SIP MESSAGE method support
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
2013 Nov 27
0
Asterisk is delaying DTMF INFO in meetme
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 at xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060 From: <sip:xxx at xxx>;tag=43 To: <sip:xxx at xxx>;tag=9753.0207 Call-ID: xxx at xxx CSeq: 25634 INFO Content-Length: 26 Content-Type: application/dtmf-relay Signal= 2\r\n
2014 Feb 17
1
Asterisk crashes at "meetme kick all"
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
2011 May 13
0
Blocking multiple SIP registration
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
2011 Apr 26
0
play audio file to destination SIP channel on attended call transfer
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
2013 Nov 16
0
Help - DTMF relay in meetme is not reliable
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
2013 Nov 17
0
DTMF relay in meetme is not reliable
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
2003 Jun 02
1
Two * Questions
We're getting ready to ditch our hosted virtual PBX for an Asterisk solution and I've got a couple questions that probably come from experience; we're looking to host two PBXs on the same box if the PBXs are identified by different DID numbers, and I was curious if: 1. Is it possible to have duplicate extensions between the two PBXs? Eg: 555-1111x100 and 555-2222x100 on same
2013 Aug 23
3
(no subject)
I am new to R. I have a data like: x y z w p .......... m 1 10 15 20 25 30 2 11 16 21 26 31 3 12 17 18 19 20 4 51 52 53 55 67 ....... thus I have 145 rows
2007 Aug 09
1
displaying svg chart
dear railers I was attempting to display a svg chart inside a tooltip in rails using Scruffy. my browser is Firefox 2 and i am using WEBRICK. when i render inside the controller using graph.render(:size=> [255,205], :to => ''C:\xyz.svg) and serve iit via rhtml using the <embed> tag. Instead of the chart inside the tooltip i get a dialog box asking me to open it ... with