Re: [asterisk-users] internal call record
On Thursday 04 April 2019 at 17:59:07, Karsten Wemheuer wrote:
> Am Sonntag, den 10.03.2019, 12:46 +0300 schrieb Gokan Atmaca:
> >
> > Mynum: 6001 , Othernum: 6002.
> >
> > I can record as follows. But I do not enter individual records for
> > each internal required. I want to do it more smoothly with a Macro.
> >
> > Thanks.
> >
> > exten => _6001,1,NoOp()
> > exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab)
> > exten => _6001,n,Dial(SIP/6001,20)
> > exten => _6001,n,StopMixMonitor()
> > exten => _6001,n,Hangup()
>
> If You are using SIP, pay attention to media setup (option
> "directmedia" in case of chan_sip).
>
> Using directmedia the media flows from end to end not running through
> asterisk. In this case recording doesn't work.
I thought this worked the other way around:
If you set directmedia, the media flows directly between endpoints, _unless_
Asterisk has some function (such as recording, there are several others) which
prevents this being possible, in which case the media continues to flow through
Asterisk.
In other words, you have to set directmedia to make it _possible_ for the
media to bypass Asterisk, but that doesn't mean that it necessarily _does_.
Antony.
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[asterisk-users] Asterisk 16.3.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 16.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.3.0 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: Security bugs fixed in this release: --- * ASTERISK-28260 - Asterisk segfault when rtp negotiation is wrong or fails (Reported by Sotiris Ganouris) New Features made in this release: --- * ASTERISK-28267 - res_stasis: Add ability to switch applications (Reported by Benjamin Keith Ford) Bugs fixed in this release: --- * ASTERISK-27541 - app_queue: Queue paused reason was (big number) secs ago when reason is set (Reported by César BenjamÃn GarcÃa MartÃnez) * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate (Reported by Olivier Krief) * ASTERISK-28350 - manager: Stasis backed up due to locking (Reported by Joshua C. Colp) * ASTERISK-25792 - chan_sip: qualifygap bounds checking (Reported by Paul Sandys) * ASTERISK-28341 - res_config_odbc eliminates empty custom (â@â prefix) variables (Reported by Alexei Gradinari) * ASTERISK-28333 - StasisEnd event makes wrong timestamp value (Reported by sungtae kim) * ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent (Reported by Jared Hull) * ASTERISK-28332 - Variable ALTCONF ignored when service is used in Debian (Reported by Cirillo Ferreira) * ASTERISK-28314 - ARI: API changed but "apiVersion" in rest-api\resources.json did not (Reported by Stefan Repke) * ASTERISK-28335 - stasis: Make topic and maybe subscription names unique and more useful (Reported by Joshua C. Colp) * ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation (Reported by sungtae kim) * ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of 183 without SDP (Reported by Torrey Searle) * ASTERISK-28328 - MeetMe global non-admin mute is muting admins that subsequently join (Reported by Philip Mott) * ASTERISK-27964 - app_queue: ring_entry accesses nativeformats without channel lock or reference (Reported by Francisco Seratti) * ASTERISK-28168 - app_queue: Adding a blank entry into sql queue_members crashes asterisk. (Reported by Michael) * ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion script fails (Reported by Guido Weckwerth) * ASTERISK-28272 - The basic-pbx config samples don't produce a running asterisk (Reported by George Joseph) * ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302 redirect (Reported by Alex Odrov) * ASTERISK-24173 - File menuselect/menuselect_gtk.c has no license header (Reported by Jeremy Lainé) * ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces (Reported by Nikolay shakin) * ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to pjsip_wizard.conf causes crash (Reported by Jonathan Harris) * ASTERISK-28166 - app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC (Reported by Michael) * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when AOR is blocked (Reported by Ross Beer) * ASTERISK-28301 - Allow voicemail boxes to be subscribed to with a presence event package (Reported by George Joseph) * ASTERISK-28303 - res_rtp_asterisk: Interaction between smoother and DTMF can cause out of order timestamps (Reported by Torrey Searle) * ASTERISK-28302 - ARI: "Error destroying mutex" when listing all ARI applications (Reported by Stefan Repke) * ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some applications (Reported by George Joseph) * ASTERISK-28106 - Astricon Feedback: Unable to filter ARI events when GETting causes overload of events (Reported by George Joseph) * ASTERISK-28284 - switching between native_bridge and simple_bridge can cause one way audio (Reported by Torrey Searle) * ASTERISK-28251 - CI: Fix CI so it reverifies commit message changes (Reported by George Joseph) * ASTERISK-28277 - database: Add some basic logging (Reported by Joshua C. Colp) * ASTERISK-28181 - ari: Originating overwrites channel start time (Reported by sungtae kim) Improvements made in this release: --- * ASTERISK-28326 - ari: Added timestamp for some ari events. (Reported by sungtae
Re: [asterisk-users] internal call record
Hi,
Am Sonntag, den 10.03.2019, 12:46 +0300 schrieb Gokan Atmaca:
> Hello
>
> Mynum: 6001 , Othernum: 6002.
>
>
> I can record as follows. But I do not enter individual records for
> each internal
> required. I want to do it more smoothly with a Macro.
>
> Thanks.
>
> exten => _6001,1,NoOp()
> exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab)
> exten => _6001,n,Dial(SIP/6001,20)
> exten => _6001,n,StopMixMonitor()
> exten => _6001,n,Hangup()
>
If You are using SIP, pay attention to media setup (option
"directmedia" in case of chan_sip).
Using directmedia the media flows from end to end not running through
asterisk. In this case recording doesn't work.
HTH,
Karsten
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[asterisk-users] Asterisk 13.26.0 Now Available
The Asterisk Development Team would like to announce the release of Asterisk 13.26.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.26.0 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: New Features made in this release: --- * ASTERISK-28267 - res_stasis: Add ability to switch applications (Reported by Benjamin Keith Ford) Bugs fixed in this release: --- * ASTERISK-20986 - QUEUE_MEMBER 's description is inaccurate (Reported by Olivier Krief) * ASTERISK-28350 - manager: Stasis backed up due to locking (Reported by Joshua C. Colp) * ASTERISK-25792 - chan_sip: qualifygap bounds checking (Reported by Paul Sandys) * ASTERISK-28341 - res_config_odbc eliminates empty custom (â@â prefix) variables (Reported by Alexei Gradinari) * ASTERISK-28333 - StasisEnd event makes wrong timestamp value (Reported by sungtae kim) * ASTERISK-28306 - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be sent (Reported by Jared Hull) * ASTERISK-27964 - app_queue: ring_entry accesses nativeformats without channel lock or reference (Reported by Francisco Seratti) * ASTERISK-28314 - ARI: API changed but "apiVersion" in rest-api\resources.json did not (Reported by Stefan Repke) * ASTERISK-28335 - stasis: Make topic and maybe subscription names unique and more useful (Reported by Joshua C. Colp) * ASTERISK-28321 - res_rtp_asterisk: Fixing possible divide by zero for rtcp stat calculation (Reported by sungtae kim) * ASTERISK-28332 - Variable ALTCONF ignored when service is used in Debian (Reported by Cirillo Ferreira) * ASTERISK-28322 - chan_pjsip: Add option to allow ignoring of 183 without SDP (Reported by Torrey Searle) * ASTERISK-28328 - MeetMe global non-admin mute is muting admins that subsequently join (Reported by Philip Mott) * ASTERISK-28168 - app_queue: Adding a blank entry into sql queue_members crashes asterisk. (Reported by Michael) * ASTERISK-28323 - pjsip: sip.conf to pjsip.conf conversion script fails (Reported by Guido Weckwerth) * ASTERISK-28272 - The basic-pbx config samples don't produce a running asterisk (Reported by George Joseph) * ASTERISK-28312 - res_pjsip_diversion: Corrupted SIP Diversion field after handling a 302 redirect (Reported by Alex Odrov) * ASTERISK-24173 - File menuselect/menuselect_gtk.c has no license header (Reported by Jeremy Lainé) * ASTERISK-28166 - app_voicemail: Asterisk unresponsive after changing voicemail password with ODBC (Reported by Michael) * ASTERISK-28309 - res_pjsip: Wrong Contact and Via fields with multiple UDP interfaces (Reported by Nikolay shakin) * ASTERISK-27992 - PJSIP: Adding `sends_registrations = yes` to pjsip_wizard.conf causes crash (Reported by Jonathan Harris) * ASTERISK-28213 - res_pjsip: Threads pile up needlessly when AOR is blocked (Reported by Ross Beer) * ASTERISK-28301 - Allow voicemail boxes to be subscribed to with a presence event package (Reported by George Joseph) * ASTERISK-28303 - res_rtp_asterisk: Interaction between smoother and DTMF can cause out of order timestamps (Reported by Torrey Searle) * ASTERISK-28302 - ARI: "Error destroying mutex" when listing all ARI applications (Reported by Stefan Repke) * ASTERISK-28300 - AST_PBX_MAX_STACK is too low for some applications (Reported by George Joseph) * ASTERISK-28106 - Astricon Feedback: Unable to filter ARI events when GETting causes overload of events (Reported by George Joseph) * ASTERISK-28284 - switching between native_bridge and simple_bridge can cause one way audio (Reported by Torrey Searle) * ASTERISK-28288 - Resources (udptl fd) leaking for T.38 calls (Reported by Paulo Vicentini) * ASTERISK-28251 - CI: Fix CI so it reverifies commit message changes (Reported by George Joseph) * ASTERISK-28277 - database: Add some basic logging (Reported by Joshua C. Colp) * ASTERISK-28181 - ari: Originating overwrites channel start time (Reported by sungtae kim) Improvements made in this release: --- * ASTERISK-28326 - ari: Added timestamp for some ari events. (Reported by sungtae kim) * ASTERISK-28317 - Add logical group at DAHDIChannel event and create "dahdi_group" at CHANNEL function (Reported by Cirillo Ferreira) * ASTERISK-28279 - Added creation timestamp for bridge (Reported by sungtae kim)
Re: [asterisk-users] PJSIP Delay in Dialing
Thanks Joshua. Hopefully I'll be able to retry tomorrow. On Thu, 4 Apr 2019 at 15:30, Joshua C. Colp wrote: > On Thu, Apr 4, 2019, at 11:27 AM, Mark Farmer wrote: > > Thanks, I did enable debugging but didn't see any attempts to resolve > > hostnames. I will give it another look. > > > > I did have an empty resolver_unbound.conf (not even a general context) > > - would that likely cause issues? I would expect the defaults to kick > > in but I have now added: > > > > [general] > > hosts=system > > > > I will retest/debug when ASAP. > > The default would have indeed kicked in, and it would have ignored > /etc/hosts and tried to do resolution using the normal resolution process. > I'd suggest providing console output somewhere so we can see precisely what > is going on. > > -- > 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 -- Mark Farmer farm...@gmail.com -- _ -- 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 Delay in Dialing
On Thu, Apr 4, 2019, at 11:27 AM, Mark Farmer wrote: > Thanks, I did enable debugging but didn't see any attempts to resolve > hostnames. I will give it another look. > > I did have an empty resolver_unbound.conf (not even a general context) > - would that likely cause issues? I would expect the defaults to kick > in but I have now added: > > [general] > hosts=system > > I will retest/debug when ASAP. The default would have indeed kicked in, and it would have ignored /etc/hosts and tried to do resolution using the normal resolution process. I'd suggest providing console output somewhere so we can see precisely what is going on. -- 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] PJSIP Delay in Dialing
Thanks, I did enable debugging but didn't see any attempts to resolve hostnames. I will give it another look. I did have an empty resolver_unbound.conf (not even a general context) - would that likely cause issues? I would expect the defaults to kick in but I have now added: [general] hosts=system I will retest/debug when ASAP. Mark. On Thu, 4 Apr 2019 at 15:20, Joshua C. Colp wrote: > On Thu, Apr 4, 2019, at 11:18 AM, Mark Farmer wrote: > > Seems to be res_resolver_unbound.so > > Reading the documentation now but any hints greatly appreciated! > > The general section of the resolver_unbound.conf file has various options, > including for using the /etc/hosts file. Per my previous email though if > you bump up the debug it'll tell you precisely what it's doing (resolving > hostname blah for an A record, etc). > > -- > 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 -- Mark Farmer farm...@gmail.com -- _ -- 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 Delay in Dialing
On Thu, Apr 4, 2019, at 11:18 AM, Mark Farmer wrote: > Seems to be res_resolver_unbound.so > Reading the documentation now but any hints greatly appreciated! The general section of the resolver_unbound.conf file has various options, including for using the /etc/hosts file. Per my previous email though if you bump up the debug it'll tell you precisely what it's doing (resolving hostname blah for an A record, etc). -- 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] PJSIP Delay in Dialing
Seems to be res_resolver_unbound.so Reading the documentation now but any hints greatly appreciated! Mark. On Thu, 4 Apr 2019 at 15:07, Joshua C. Colp wrote: > On Thu, Apr 4, 2019, at 11:03 AM, Mark Farmer wrote: > > Sorry, should have included that. > > > > Asterisk 16.2.1 > > And what res_resolver module is loaded and in use? Depending on the module > it may not be using /etc/hosts. You can also increase debug (debug to > console in logger.conf and core set debug 9) and the resolver will tell you > what it is trying to do. > > -- > 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 -- Mark Farmer farm...@gmail.com -- _ -- 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 Delay in Dialing
On Thu, Apr 4, 2019, at 11:03 AM, Mark Farmer wrote: > Sorry, should have included that. > > Asterisk 16.2.1 And what res_resolver module is loaded and in use? Depending on the module it may not be using /etc/hosts. You can also increase debug (debug to console in logger.conf and core set debug 9) and the resolver will tell you what it is trying to do. -- 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] PJSIP Delay in Dialing
Sorry, should have included that. Asterisk 16.2.1 Mark. On Thu, 4 Apr 2019 at 14:56, Joshua C. Colp wrote: > On Thu, Apr 4, 2019, at 10:53 AM, Mark Farmer wrote: > > As I understand it, delays like this are almost always caused by slow > > or failing DNS lookups. Running a packet capture on all interfaces > > filtering on port 53 shows no DNS traffic leaving the server. I have > > ensured that there is a DNS record for the server & that it can resolve > > it. I've also added records to my hosts file and checked using 'genet > > ahosts hostname' but still the issue remains. > > > > So how do I figure out what is going wrong please? This is preventing > > me from moving from chan_sip to chan_pjsip. > > What version of Asterisk? That will change the answer as 13 uses the > built-in PJSIP DNS resolver, while 16 uses our own implementation. > > -- > 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 -- Mark Farmer farm...@gmail.com -- _ -- 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 Delay in Dialing
On Thu, Apr 4, 2019, at 10:53 AM, Mark Farmer wrote: > As I understand it, delays like this are almost always caused by slow > or failing DNS lookups. Running a packet capture on all interfaces > filtering on port 53 shows no DNS traffic leaving the server. I have > ensured that there is a DNS record for the server & that it can resolve > it. I've also added records to my hosts file and checked using 'genet > ahosts hostname' but still the issue remains. > > So how do I figure out what is going wrong please? This is preventing > me from moving from chan_sip to chan_pjsip. What version of Asterisk? That will change the answer as 13 uses the built-in PJSIP DNS resolver, while 16 uses our own implementation. -- 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
[asterisk-users] PJSIP Delay in Dialing
As I understand it, delays like this are almost always caused by slow or failing DNS lookups. Running a packet capture on all interfaces filtering on port 53 shows no DNS traffic leaving the server. I have ensured that there is a DNS record for the server & that it can resolve it. I've also added records to my hosts file and checked using 'genet ahosts hostname' but still the issue remains. So how do I figure out what is going wrong please? This is preventing me from moving from chan_sip to chan_pjsip. Many thanks Mark. -- _ -- 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] Message: Authentication failed on manager interface
On Thu, 2019-04-04 at 15:08 +0200, Antony Stone wrote: > > It's not "Password", it's "Secret" :) Ha ha. I knew it would be a head-smack type problem. Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- 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] Message: Authentication failed on manager interface
On Thursday 04 April 2019 at 14:28:15, Brian J. Murrell wrote: > # echo -e "Action: Login\r\nUsername: myasterisk\r\nPassword: a\r\n\r\n" It's not "Password", it's "Secret" :) Antony. -- I don't know, maybe if we all waited then cosmic rays would write all our software for us. Of course it might take a while. - Ron Minnich, Los Alamos National Laboratory 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] Message: Authentication failed on manager interface
I'm not sure how much more simple I can make this but I just cannot seem to get my Asterisk 13 to accept a connection on the manager interface: --- manager.conf --- [general] enabled = yes port = 5038 bindaddr = 127.0.0.1 [myasterisk] secret=a permit=0.0.0.0/0.0.0.0 read = all write = all So, couldn't be any more wide open and simpler to connect yet: # echo -e "Action: Login\r\nUsername: myasterisk\r\nPassword: a\r\n\r\n" | ncat 127.0.0.1 5038 Asterisk Call Manager/2.10.4 Response: Error Message: Authentication failed Even more basically if you like: # telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk Call Manager/2.10.4 Action: Login Username: myasterisk Password: a Response: Error Message: Authentication failed Connection closed by foreign host. I have restarted asterisk after editing manager.conf, even though a manager reload should be all that's needed. server*CLI> manager show user myasterisk username: myasterisk secret: ACL: yes read perm: system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,security,message,all write perm: system,call,log,verbose,command,agent,user,config,dtmf,reporting,cdr,dialplan,originate,agi,cc,aoc,test,security,message,all displayconnects: yes allowmultiplelogin: yes Variables: I know this is going to be a "head smack" problem, but seriously. What am I missing? Cheers, b. signature.asc Description: This is a digitally signed message part -- _ -- 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] Asterisk13 - Dialplan reload does not take modification in account
Hi all, I switched an old asterisk 1.8 to a new 13 version, stock version from Ubuntu 18.04 server. I did some modification in dialplan but after a reload they are not taken in account :(, even after restarting asterisk. I checked logs and found lots of merging incls/swits/igpats from old(localEP) to new(localEP) context, registrar = pbx_lua also lines ending with pbx_config. What does this mean? Could be the source of my troubles ? Thanks for any hint Daniel -- _ -- 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
