Re: [asterisk-users] MulticastRTP and ttl
On 5/12/2021 9:37 AM, Jerry Geis wrote: > Sorry it - may have worked - my person only used a single / not // > Thanks! > > Does this work on version 13 or just version 18 ? In terms of supported versions of Asterisk it works in 16+ Kind regards, Sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MulticastRTP and ttl
On 5/12/2021 9:19 AM, Jerry Geis wrote: > I tried the 239.1.2.3:20480//t(5) and still using a default of 1. > Is there a config file to set this default TTL ? No, just the syntax I already suggested. It's documented here: https://wiki.asterisk.org/wiki/display/AST/New+in+14#Newin14-chan_rtp(waschan_multicast_rtp) Kind regards, Sean-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MulticastRTP and ttl
On Wed, May 12, 2021 at 9:19 AM Jerry Geis wrote: > > On Tue, May 11, 2021 at 4:24 PM Jerry Geis wrote: > >> Hi - >> >> I was using asterisk 13.36.0 and tried to specify a MulticastRTP TTL with >> Channel: MulticastRTP/basic/239.1.2.3:20480/5 >> where 5 is the ttl >> >> This did not work. >> So I updated to asterisk 18.4.0 tried the same thing >> and did not work. >> >> I remove the /5 and just do regular and it works but the ttl is 1 - I >> need 5. >> >> How can I get ttl on multicastrtp ? >> >> >> Thanks >> >> Jerry >> > > I tried the 239.1.2.3:20480//t(5) and still using a default of 1. > Is there a config file to set this default TTL ? > > Jerry > Sorry it - may have worked - my person only used a single / not // Thanks! Does this work on version 13 or just version 18 ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MulticastRTP and ttl
On Tue, May 11, 2021 at 4:24 PM Jerry Geis wrote: > Hi - > > I was using asterisk 13.36.0 and tried to specify a MulticastRTP TTL with > Channel: MulticastRTP/basic/239.1.2.3:20480/5 > where 5 is the ttl > > This did not work. > So I updated to asterisk 18.4.0 tried the same thing > and did not work. > > I remove the /5 and just do regular and it works but the ttl is 1 - I need > 5. > > How can I get ttl on multicastrtp ? > > > Thanks > > Jerry > I tried the 239.1.2.3:20480//t(5) and still using a default of 1. Is there a config file to set this default TTL ? Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MulticastRTP and ttl
On 5/11/2021 4:24 PM, Jerry Geis wrote: > I was using asterisk 13.36.0 and tried to specify a MulticastRTP TTL with > Channel: MulticastRTP/basic/239.1.2.3:20480/5 > where 5 is the ttl Try: MulticastRTP/basic/239.1.2.3:20480//t(5) Kind regards, Sean-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MulticastRTP and ttl
Hi - I was using asterisk 13.36.0 and tried to specify a MulticastRTP TTL with Channel: MulticastRTP/basic/239.1.2.3:20480/5 where 5 is the ttl This did not work. So I updated to asterisk 18.4.0 tried the same thing and did not work. I remove the /5 and just do regular and it works but the ttl is 1 - I need 5. How can I get ttl on multicastrtp ? Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multicastRTp
On 2014-08-08 21:54, Jerry Geis wrote: On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis ge...@pagestation.com mailto:ge...@pagestation.com wrote: I am using a cyberdata sip paging adapter and with the Dial(MulticastRTP/basic/IP:port) and with tshark I see the RTP data, my device looks like its accepting the data and I hear a click for my relay on my device so it would seem its accepting the call, however - I hear no audio... If I call using the dial plan everything seems to work... Is there an issue with using call files ? Channel: MulticastRTP/basic/239.168.3.10:11000 http://239.168.3.10:11000 It all seems to work, I see multicast audio, the unit answers, I just get no audio or crappy audio... Is the codec not set right in that case from a call file? How do I set the codec for multicastrtp in a call file? might make sense that speak live the codec is already established but from a call file there is no codec Any thoughts or how do I set the codec in a call file for multicast to try it? Please check this link and see if this applies to you: http://www.voip-info.org/wiki/view/Asterisk+MulticastRTP+channels Regards Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multicastRTp
On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis ge...@pagestation.com wrote: I am using a cyberdata sip paging adapter and with the Dial(MulticastRTP/basic/IP:port) and with tshark I see the RTP data, my device looks like its accepting the data and I hear a click for my relay on my device so it would seem its accepting the call, however - I hear no audio... Asterisk 11.11.0 is what I am using. What might be wrong here? Thanks, jerry If I call using the dial plan everything seems to work... Is there an issue with using call files ? Channel: MulticastRTP/basic/239.168.3.10:11000 It all seems to work, I see multicast audio, the unit answers, I just get no audio or crappy audio... Is the codec not set right in that case from a call file? How do I set the codec for multicastrtp in a call file? might make sense that speak live the codec is already established but from a call file there is no codec Any thoughts or how do I set the codec in a call file for multicast to try it? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multicastRTp
I am using a cyberdata sip paging adapter and with the Dial(MulticastRTP/basic/IP:port) and with tshark I see the RTP data, my device looks like its accepting the data and I hear a click for my relay on my device so it would seem its accepting the call, however - I hear no audio... Asterisk 11.11.0 is what I am using. What might be wrong here? Thanks, jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multicastRTP source interface
I have an asterisk 11.4.0 server with two interfaces, eth0 and eth1. Eth0 has a default gateway on it, eth1 is connected the subnet that has my phones registered. I'd like to use the multicastRTP driver to do paging. However, when a phone dials an extension with multicastRTP, the multicast stream goes to the primary interface (eth0) and it really needs to go to eth1. Is there a way to specify which interface the rtp is sourced from? Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
