[asterisk-users] Music on hold depending on who put call on hold
Hello, Does anyone know of a way to play different music on hold depending on which party puts the call on hold? We can specify the music on hold per channel, but that doesn't do what is needed. We want to play one music if the caller puts the call on hold, and a different music if the called party puts the call on hold. Thanks in advance for any assistance. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _ -- 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] PJSIP Setup Outbound SIP Trunk
On Mon, Oct 14, 2019 at 11:56 AM Ahmed Chohan wrote: > Hi, > > I've currently migrating from chan_sip to chan_pjsip, for now I'm able to > setup and configured extensions in PJSIP and incoming trunks but unable to > configure outbound trunk as getting unauth/unregistered trunk endpoint > message error message when making outbound calls. However, for inbound > calls I'm not facing any issues. > > I would like to know how can I configured outbound sip trunk bypassing > registration and auth? > Where are the messages coming from? Is Asterisk sending an outbound registration, but getting rejected? If so make sure your username/password credentials are correct. > > See below current configuration; > > [trunk_proxy] > type=endpoint > transport=transport-udp > context=fromsip > disallow=all > allow=ulaw > aors=trunk_proxy > force_rport=no > direct_media=yes > ice_support=no > trust_id_inbound=yes > outbound_auth=trunk_proxy > > [trunk_proxy] > type=aor > contact=sip:10.3.120.208:5060 > > [trunk_proxy] > type=identify > endpoint=trunk_proxy > match=10.3.120.208 > > [trunk_proxy] > type=auth > auth_type=userpass > password= > username=sip_proxy > > [trunk_proxy] > type=registration > outbound_auth=trunk_proxy > server_uri=sip:10.3.120.208:5060 > client_uri=sip:10.3.120.208:5060 > auth_rejection_permanent=no > > -- > Regards, > > Ahmed Munir Chohan > > -- > _ > -- 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 -- Kevin Harwell Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: https://digium.com & https://asterisk.org -- _ -- 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] [asterisk-app-dev] ARI Channel recording
On Wed, Oct 16, 2019, at 3:03 PM, Marcelo Garay wrote: > Thank you for your answer!! > > Unfortunately I'm using the CEF browser based on Chromium and it > doesn't support H264 because license isn't free so renegotiation is not > an option. > > I've noticed when recording a channel with video asterisk automatically > tries to save the video feed to a separate file besides the .wav. In my > case I can see "file.c:1484 ast_writefile: No such format 'vp9' " error > in the logs, so I would assume is just that the code for VP9 encoding > hasn't been added to Asterisk yet. Do you know if this is due to any > other reason besides nobody taking the time to implement it (reasons > like VP9 licensing, performance hit, etc.)? It seems like VP9 is > royalty-free and the encoder source code is on GitHub. I might try to > look into making a PR for this sometime in the future if I have some > time, but I don't want to waste my time if this idea has already been > discussed among developers and discarded for some reason. Encoding is not the same as file recording and playback. It's how the data is stored in a file and retrieved, which doesn't involve any conversion. I don't think anyone has discussed working on such a thing or thought about it really. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org ___ asterisk-app-dev mailing list asterisk-app-...@lists.digium.com http://lists.digium.com/cgi-bin/mailman/listinfo/asterisk-app-dev -- _ -- 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 16.6.1 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 16.6.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.6.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: --- * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) * ASTERISK-28575 - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.1 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 13.29.1 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 13.29.1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.29.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release: Bugs fixed in this release: --- * ASTERISK-28574 - pjproject fails to build on 16.6.0, works on 16.5 (Reported by Niklas Larsson) * ASTERISK-28575 - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson) For a full list of changes in this release, please see the ChangeLog: https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.29.1 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] clarification on gosub, macros and AEL
Hello Joshua, Wednesday, October 16, 2019, 10:39:27 AM, you wrote: > The module is still present, it just isn't built by default. It > requires explicit enabling using menuselect. I was obviously too concerned about the rebuilding of wanpipe with a new kernel version to notice anything about macros in menuselect! As I said, it is not hard to convert from macro to gosub, once I had worked out why my dialplan wasn't working. -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- 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] clarification on gosub, macros and AEL
On Wed, Oct 16, 2019, at 6:28 AM, Julian Beach wrote: > Hello Doug, > > Tuesday, October 15, 2019, 5:07:45 PM, you wrote: > > > Personally, I don't think MACROS are going anywhere any time soon, > > so I have not bothered looking into a substitution. > > Haven't they gone in Asterisk 16? I've just upgraded from Asterisk 13 > and had to do some hasty re-writing of my dialplan to convert the > Macros into Gosubs because the Macros did not work (I had noticed the > deprecation notice previously, but forgot about it). I didn't have > many, so I did them by hand, but it would be easy to script. The module is still present, it just isn't built by default. It requires explicit enabling using menuselect. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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] clarification on gosub, macros and AEL
Hello Doug, Tuesday, October 15, 2019, 5:07:45 PM, you wrote: > Personally, I don't think MACROS are going anywhere any time soon, > so I have not bothered looking into a substitution. Haven't they gone in Asterisk 16? I've just upgraded from Asterisk 13 and had to do some hasty re-writing of my dialplan to convert the Macros into Gosubs because the Macros did not work (I had noticed the deprecation notice previously, but forgot about it). I didn't have many, so I did them by hand, but it would be easy to script. Julian -- Best regards, Julianmailto:jb_s...@trink.co.uk -- _ -- 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
