[asterisk-users] Asterisk 17.5.1 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 17.5.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.5.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following issue is resolved in this release: Bugs fixed in this release: --- * ASTERISK-28948 - ARI channel create doesn't referencing the channel_id parameter (Reported by sungtae kim) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1 Thank you for your continued support of Asterisk! -- _ -- 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] Asterisk 16.11.1 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 16.11.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.11.1 resolves an issue reported by the community and would have not been possible without your participation. Thank you! The following issue is resolved in this release: Bugs fixed in this release: --- * ASTERISK-28948 - ARI channel create doesn't referencing the channel_id parameter (Reported by sungtae kim) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1 Thank you for your continued support of Asterisk! -- _ -- 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] Voice "broken" during calls
On 6/16/20 1:18 AM, Luca Bertoncello wrote: Am 15.06.2020 23:15, schrieb Jeff LaCoursiere: Hi again, just a question, to be sure... sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap & sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap & eth0 is my DSL interface and eth1 my phone interface? Sure, that's fine. We will figure out which one is north/south in the analysis. Try to limit the traffic to just your phone call tests (to reduce the size of the capture files). Make all your tests, then: Well, assuming eth0 is the DSL interface and eth1 the phone interface, I can so that: tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de & tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.xx (IP of my phone) & is it correct? Perfect. Cheers, j -- _ -- 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] Voice "broken" during calls
Am 16.06.2020 10:48, schrieb Antony Stone: On Tuesday 16 June 2020 at 08:18:51, Luca Bertoncello wrote: > sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap & > sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap & eth0 is my DSL interface and eth1 my phone interface? Well, one is internal (phone) and the other is external (DT), doesn't matter which way round. This was what I meant... tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de & tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.xx (IP of my phone) & Looks like you name your Banana interfaces very similarly to mine :) I think, we are not alone... :D However, I would be careful with that first one, containing "host tel.t- online.de". I don't use DT, so I can't be sure, but I guess this is the SIP server to which you register with the account credentials... It *may not* be the same machine as handles the RTP packets - that is negotiated separately between Asterisk (or the Thomson, when it's connected directly to DT) as part of the SIP INVITE / Acknowledge. So, you *could* find that you capture all of the SIP traffic and none of the RTP traffic. On the other hand, you might get everything. You can be pretty sure it's worked if you do the above and then find that the two packet capture files are approximately the same size. If the DT one is significantly smaller (by which I mean a factor of at least ten different), then omit the "host" parameter on that capture and try again... OK, I'll check it... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Voice "broken" during calls
On Tuesday 16 June 2020 at 08:18:51, Luca Bertoncello wrote: > > sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap & > > sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap & > > eth0 is my DSL interface and eth1 my phone interface? Well, one is internal (phone) and the other is external (DT), doesn't matter which way round. > tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de & > tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.xx (IP of my > phone) & Looks like you name your Banana interfaces very similarly to mine :) However, I would be careful with that first one, containing "host tel.t- online.de". I don't use DT, so I can't be sure, but I guess this is the SIP server to which you register with the account credentials... It *may not* be the same machine as handles the RTP packets - that is negotiated separately between Asterisk (or the Thomson, when it's connected directly to DT) as part of the SIP INVITE / Acknowledge. So, you *could* find that you capture all of the SIP traffic and none of the RTP traffic. On the other hand, you might get everything. You can be pretty sure it's worked if you do the above and then find that the two packet capture files are approximately the same size. If the DT one is significantly smaller (by which I mean a factor of at least ten different), then omit the "host" parameter on that capture and try again... Antony. -- A few words to be cautious of between American and English: - momentarily - suspenders - chips - pants - jelly - pavement - vest - pint (and gallon) - pissed Please reply to the list; please *don't* CC me. -- _ -- 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] Asterisk conference manager
Hi, I'm looking for some software to allow users to login and manage Asterisk conferences (define conference user lists, call users to join a conference, etc.). I'm aware of astconfman (https://github.com/litnimax/astconfman). Are there any others? Thanks in advance. Regards, IanG -- _ -- 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
