[asterisk-users] asterisk release 21.1.0
to be escaped. Ampersands in URLs passed to the `Playback()`, `Background()`, `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()` applications as filename arguments can now be escaped by single quoting the filename. Additionally, this is also possible when using the `CONFBRIDGE` dialplan function, or configuring various features in `confbridge.conf` and `queues.conf`. - ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. The dtls_rekey will be disabled if webrtc support is requested on an endpoint. A warning will also be emitted. - ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters As part of this update, the maximum allowable length for PJSIP endpoints and relevant resources has been increased from 40 to 255 characters. To take advantage of this enhancement, it is recommended to run the necessary procedures (e.g., Alembic) to update your schemas. Closed Issues: - #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4 - #129: [bug]: res_speech_aeap: Crash due to NULL format on setup - #242: [new-feature]: logger: Allow filtering logs in CLI by channel - #248: [bug]: core_local: Local channels cannot have slashes in the destination - #260: [bug]: maxptime must be changed to multiples of 20 - #286: [improvement]: chan_iax2: Improve authentication debugging - #289: [new-feature]: Add support for deleting channel and global variables - #294: [improvement]: chan_dahdi: Improve call pickup documentation - #298: [improvement]: chan_rtp: Implement RTP glue - #301: [bug]: Number of ICE TURN threads continually growing - #303: [bug]: SpeechBackground never exits - #308: [bug]: chan_console: Deadlock when hanging up console channels - #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/ - #316: [bug]: Privilege Escalation in Astrisk's Group permissions. - #319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel - #321: [bug]: Performance suffers unnecessarily when debugging deadlocks - #325: [bug]: hangup after beep to avoid waiting for timeout - #330: [improvement]: Add cel user event helper function - #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in build_resource_tree - #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases - #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If - #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality - #349: [improvement]: Add libjwt to third-party - #352: [bug]: Update qualify_timeout documentation to include DNS note - #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line - #356: [new-feature]: app_directory: Add ADSI support. - #360: [improvement]: Update documentation for CHANGES/UPGRADE files - #362: [improvement]: Speed up ARI command processing - #379: [bug]: Orphaned taskprocessors cause shutdown delays - #384: [bug]: Unnecessary re-INVITE after answer - #388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats - #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided - #398: [new-feature]: app_voicemail: Add AMI event for password change - #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence - #423: [improvement]: func_lock: Add missing see-also refs - #425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample - #428: [bug]: cli: Output is truncated from "config show help" - #430: [bug]: Fix broken links - #442: [bug]: func_channel: Some channel options are not settable - #445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases - #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO - #462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller - #465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations - #482: [improvement]: manager.c: Improve clarity of "manager show connected" output - #509: [bug]: res_pjsip: Crash when looking up transport state in use - #513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG - #520: [improvement]: menuselect: Use more specific error message. - #530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels - #539: [bug]: Existence of logger.xml causes linking failure -- _ -- 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
[asterisk-users] asterisk release 20.6.0
ngle quoting the filename. Additionally, this is also possible when using the `CONFBRIDGE` dialplan function, or configuring various features in `confbridge.conf` and `queues.conf`. - ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. The dtls_rekey will be disabled if webrtc support is requested on an endpoint. A warning will also be emitted. - ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters As part of this update, the maximum allowable length for PJSIP endpoints and relevant resources has been increased from 40 to 255 characters. To take advantage of this enhancement, it is recommended to run the necessary procedures (e.g., Alembic) to update your schemas. Closed Issues: - #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4 - #129: [bug]: res_speech_aeap: Crash due to NULL format on setup - #242: [new-feature]: logger: Allow filtering logs in CLI by channel - #248: [bug]: core_local: Local channels cannot have slashes in the destination - #260: [bug]: maxptime must be changed to multiples of 20 - #286: [improvement]: chan_iax2: Improve authentication debugging - #289: [new-feature]: Add support for deleting channel and global variables - #294: [improvement]: chan_dahdi: Improve call pickup documentation - #298: [improvement]: chan_rtp: Implement RTP glue - #301: [bug]: Number of ICE TURN threads continually growing - #303: [bug]: SpeechBackground never exits - #308: [bug]: chan_console: Deadlock when hanging up console channels - #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/ - #316: [bug]: Privilege Escalation in Astrisk's Group permissions. - #319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel - #321: [bug]: Performance suffers unnecessarily when debugging deadlocks - #325: [bug]: hangup after beep to avoid waiting for timeout - #330: [improvement]: Add cel user event helper function - #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases - #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If - #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality - #349: [improvement]: Add libjwt to third-party - #352: [bug]: Update qualify_timeout documentation to include DNS note - #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line - #356: [new-feature]: app_directory: Add ADSI support. - #360: [improvement]: Update documentation for CHANGES/UPGRADE files - #362: [improvement]: Speed up ARI command processing - #379: [bug]: Orphaned taskprocessors cause shutdown delays - #384: [bug]: Unnecessary re-INVITE after answer - #388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats - #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided - #398: [new-feature]: app_voicemail: Add AMI event for password change - #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence - #423: [improvement]: func_lock: Add missing see-also refs - #425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample - #428: [bug]: cli: Output is truncated from "config show help" - #430: [bug]: Fix broken links - #442: [bug]: func_channel: Some channel options are not settable - #445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases - #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO - #462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller - #465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations - #482: [improvement]: manager.c: Improve clarity of "manager show connected" output - #509: [bug]: res_pjsip: Crash when looking up transport state in use - #513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG - #520: [improvement]: menuselect: Use more specific error message. - #530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels - #539: [bug]: Existence of logger.xml causes linking failure -- _ -- 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 release 18.21.0
filename. Additionally, this is also possible when using the `CONFBRIDGE` dialplan function, or configuring various features in `confbridge.conf` and `queues.conf`. - ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled. The dtls_rekey will be disabled if webrtc support is requested on an endpoint. A warning will also be emitted. - ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255 characters As part of this update, the maximum allowable length for PJSIP endpoints and relevant resources has been increased from 40 to 255 characters. To take advantage of this enhancement, it is recommended to run the necessary procedures (e.g., Alembic) to update your schemas. Closed Issues: - #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4 - #129: [bug]: res_speech_aeap: Crash due to NULL format on setup - #242: [new-feature]: logger: Allow filtering logs in CLI by channel - #248: [bug]: core_local: Local channels cannot have slashes in the destination - #260: [bug]: maxptime must be changed to multiples of 20 - #286: [improvement]: chan_iax2: Improve authentication debugging - #289: [new-feature]: Add support for deleting channel and global variables - #294: [improvement]: chan_dahdi: Improve call pickup documentation - #298: [improvement]: chan_rtp: Implement RTP glue - #301: [bug]: Number of ICE TURN threads continually growing - #303: [bug]: SpeechBackground never exits - #308: [bug]: chan_console: Deadlock when hanging up console channels - #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files before /var/lib/asterisk/sounds/ - #316: [bug]: Privilege Escalation in Astrisk's Group permissions. - #319: [bug]: func_periodic_hook truncates long channel names when setting EncodedChannel - #321: [bug]: Performance suffers unnecessarily when debugging deadlocks - #325: [bug]: hangup after beep to avoid waiting for timeout - #330: [improvement]: Add cel user event helper function - #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory in some cases - #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If - #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved Functionality - #349: [improvement]: Add libjwt to third-party - #352: [bug]: Update qualify_timeout documentation to include DNS note - #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line - #356: [new-feature]: app_directory: Add ADSI support. - #360: [improvement]: Update documentation for CHANGES/UPGRADE files - #362: [improvement]: Speed up ARI command processing - #379: [bug]: Orphaned taskprocessors cause shutdown delays - #384: [bug]: Unnecessary re-INVITE after answer - #388: [bug]: Crash in app_followme.c due to not acquiring a reference to nativeformats - #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many provided - #398: [new-feature]: app_voicemail: Add AMI event for password change - #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence - #423: [improvement]: func_lock: Add missing see-also refs - #425: [improvement]: configs: Improve documentation for bandwidth in iax.conf.sample - #428: [bug]: cli: Output is truncated from "config show help" - #430: [bug]: Fix broken links - #442: [bug]: func_channel: Some channel options are not settable - #445: [bug]: ast_coredumper isn't figuring out file locations properly in all cases - #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO - #462: [new-feature]: app_dial: Add new option to preserve initial stream topology of caller - #465: [improvement]: Change res_odbc connection pool request logic to not lock around blocking operations - #482: [improvement]: manager.c: Improve clarity of "manager show connected" output - #509: [bug]: res_pjsip: Crash when looking up transport state in use - #513: [bug]: manager.c: Crash due to regression using wrong free function when built with MALLOC_DEBUG - #520: [improvement]: menuselect: Use more specific error message. - #530: [bug]: bridge_channel.c: Stream topology change amplification with multiple layers of Local channels - #539: [bug]: Existence of logger.xml causes linking failure -- _ -- 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] Mailing List Shutdown Reminder
Hello, Just a reminder that on February 1st this mailing list will go into a moderated only state meaning new messages will not be accepted. Conversations should move to the community forums[1] to continue them. Archives will remain available. Cheers, [1] https://community.asterisk.org -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] aeap wss connection
On Tue, Jan 16, 2024 at 9:56 AM marek wrote: > hi, > > i'm trying asterisk AEAP through Haproxy > > > https://docs.asterisk.org/Asterisk_18_Documentation/API_Documentation/Module_Configuration/res_aeap/?h= > > backend speech-gateway-dev-wss > mode http >option forwardfor >option http-server-close >server speech localhost:9811 > > > topology > > Asterisk - Haproxy - Node.js app - Google STT > > > Asterisk - Node.js works ok > > > tests with curl/wsscat are ok > > but asterisk as wss client doesnt work > Looking at the code it doesn't appear as though it was implemented with support for it from what I can tell. -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] aeap wss connection
hi, i'm trying asterisk AEAP through Haproxy https://docs.asterisk.org/Asterisk_18_Documentation/API_Documentation/Module_Configuration/res_aeap/?h= backend speech-gateway-dev-wss mode http option forwardfor option http-server-close server speech localhost:9811 topology Asterisk - Haproxy - Node.js app - Google STT Asterisk - Node.js works ok tests with curl/wsscat are ok but asterisk as wss client doesnt work it looks like the issue is because Asterisk sending http upgrade request to HTTPS 443 port as HTTP (no TLS handshake) Hypertext Transfer Protocol [Expert Info (Warning/Security): Unencrypted HTTP protocol detected over encrypted port, could indicate a dangerous misconfiguration.] [Unencrypted HTTP protocol detected over encrypted port, could indicate a dangerous misconfiguration.] [Severity level: Warning] [Group: Security] GET / HTTP/1.1\r\n [Expert Info (Chat/Sequence): GET / HTTP/1.1\r\n] [GET / HTTP/1.1\r\n] [Severity level: Chat] [Group: Sequence] Request Method: GET Request URI: / Request Version: HTTP/1.1 Sec-WebSocket-Version: 13\r\n Upgrade: websocket\r\n Connection: Upgrade\r\n Host: speech-gateway-dev.example.com:443\r\n Sec-WebSocket-Key: MvncKwBJv2J3AA==\r\n Sec-WebSocket-Protocol: speech_to_text\r\n \r\n [Full request URI: http://speech-gateway-dev.example.com:443/] [HTTP request 1/1] what do you think? is it bug? Marek -- _ -- 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] retry loop in ansible ?
Hi,
> Am 08.01.2024 um 18:16 schrieb C. Maj :
>
> On 12/6/23 02:08, Axel Rau wrote:
>> I have a simple config with some phones ringing simultaneously.
>> Some of them are softphones (zoiper apps on iPhone w/o push notification).
>> If such an app did bot register in time, it has no chance to pick up the
>> call.
>> If I could configure a retry loop checking for registered candidates,
>> say once a second until one phone takes the call, this would allow me
>> to pick up the call with zoiper app registered late.
>> How could this be done in ansible?
>
> Did you mean asterisk ?
Yes.
>
> If so, then you might look into the While()/EndWhile() applications, combined
> with timeouts to Dial() application, starting with something very basic such
> as the following:
>
> same = n,Set(tries=0)
> same = n,While($[${INC(tries)}<99])
> same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(1234)})
> same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(5678)})
> same = n,Dial(${team},10)
> same = n,Wait(1)
> same = n,EndWhile()
>
> ...at most that would be 11 seconds in between registration of x5678 and the
> next time it gets called when x1234 is not answering.
Thanks a lot for this example.
>
> Other approaches might involve Queue()'s with some ChannelRedirect()'s or
> even Bridge()'s, maybe AGI/ARI, etc.
>
> BTW the Asterisk Forums are a great place to post these kinds of questions in
> the future: https://community.asterisk.org
I will try this in the future,
Regards, Axel
—
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Re: [asterisk-users] retry loop in ansible ?
On 12/6/23 02:08, Axel Rau wrote:
I have a simple config with some phones ringing simultaneously.
Some of them are softphones (zoiper apps on iPhone w/o push notification).
If such an app did bot register in time, it has no chance to pick up the call.
If I could configure a retry loop checking for registered candidates,
say once a second until one phone takes the call, this would allow me
to pick up the call with zoiper app registered late.
How could this be done in ansible?
Did you mean asterisk ?
If so, then you might look into the While()/EndWhile() applications,
combined with timeouts to Dial() application, starting with something
very basic such as the following:
same = n,Set(tries=0)
same = n,While($[${INC(tries)}<99])
same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(1234)})
same = n,Set(PUSH(team,&)=${PJSIP_DIAL_CONTACTS(5678)})
same = n,Dial(${team},10)
same = n,Wait(1)
same = n,EndWhile()
...at most that would be 11 seconds in between registration of x5678 and
the next time it gets called when x1234 is not answering.
Other approaches might involve Queue()'s with some ChannelRedirect()'s
or even Bridge()'s, maybe AGI/ARI, etc.
BTW the Asterisk Forums are a great place to post these kinds of
questions in the future: https://community.asterisk.org
Regards,
--
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Penguin PBX Solutions
Denver 720-32-42-72-9
Beyond 1-833-PNGN-PBX
http://PeNGuiNPBX.com
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Re: [asterisk-users] SIP_HEADER GET_TRANSFERRER_DATA chan_pjsip
On Mon, Jan 8, 2024 at 12:07 PM marek wrote: > hi, > > we are moving our asterisk from chan_sip to chan_pjsip > > we are using SIP_HEADER with GET_TRANSFERRER_DATA for headers from > REFER (asterisk - other pbbx - SIP REFER - asterisk) > > > https://github.com/ca4ti/asterisk/commit/4b58609c331c013845a0a61d946cbbc82092170e > > is it supported in pjsip too? or is there other way? > Looking at the REFER implementation[1] it seems like no. You can submit a feature request here[2]. [1] https://github.com/asterisk/asterisk/blob/20/res/res_pjsip_refer.c [2] https://github.com/asterisk/asterisk-feature-requests -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] SIP_HEADER GET_TRANSFERRER_DATA chan_pjsip
hi, we are moving our asterisk from chan_sip to chan_pjsip we are using SIP_HEADER with GET_TRANSFERRER_DATA for headers from REFER (asterisk - other pbbx - SIP REFER - asterisk) https://github.com/ca4ti/asterisk/commit/4b58609c331c013845a0a61d946cbbc82092170e is it supported in pjsip too? or is there other way? thanks Marek -- _ -- 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] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
On 1/3/24 04:53, Henning Follmann wrote:
On Jan 2, 2024, at 23:17, the...@sys-concept.com wrote:
On 1/2/24 15:13, aster...@phreaknet.org wrote:
On 1/2/2024 4:03 PM, the...@sys-concept.com wrote:
I'm using asterisk-16.30.1
When I try to call another asterisk server over IAX I get a busy signal,
chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
-- IAX2/192.168.143.1:4569-656 is circuit-busy
Asterisk-16.16 is working normally, no congestion error.
There is not enough information for anyone to really help or comment on this.
Dialplan and IAX2 configuration on both sides of the trunk?
CLI output on both sides with iax2 debug enabled?
It is very simple:
Local Asterisk, iax.conf:
[clinic_server]
type=friend
host=dynamic
context=internal
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=no
callgroup=1
pickupgroup=1
extension.conf:
exten => 4,1,Dial(IAX2/home_server:@${clinic_server}/${EXTEN},60,rw)
exten => 4,n,Hangup() Remote Asterisk iax.conf:
[home_server]
type=friend
host=dynamic
secret=
context=extensions
disallow=all
allow=ulaw
allow=alaw
callgroup=1
pickupgroup=1
Remote extension.conf:
exten => 4,1,Dial(SIP/4,15,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(SIP/54,20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
You have no internal context in your dialplan. But in your iax.conf you specify
internal as your context.
-H
I forgot to write, the remote asterisk has:
[internal]
...
include => extensions
[extensions]
exten => 4,1,Dial(SIP/4,15,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(SIP/54,20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
So on the remote asterisk there is context [internal]
I even noticed starting asterisk-16.30 is much slower than starting
Asterisk-16.16
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Re: [asterisk-users] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
> On Jan 2, 2024, at 23:17, the...@sys-concept.com wrote:
>
> On 1/2/24 15:13, aster...@phreaknet.org wrote:
>>> On 1/2/2024 4:03 PM, the...@sys-concept.com wrote:
>>> I'm using asterisk-16.30.1
>>>
>>> When I try to call another asterisk server over IAX I get a busy signal,
>>>
>>> chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
>>> -- IAX2/192.168.143.1:4569-656 is circuit-busy
>>>
>>> Asterisk-16.16 is working normally, no congestion error.
>> There is not enough information for anyone to really help or comment on this.
>> Dialplan and IAX2 configuration on both sides of the trunk?
>> CLI output on both sides with iax2 debug enabled?
>
> It is very simple:
>
> Local Asterisk, iax.conf:
>
> [clinic_server]
> type=friend
> host=dynamic
> context=internal
> disallow=all
> allow=ulaw
> allow=alaw
> requirecalltoken=no
> callgroup=1
> pickupgroup=1
>
> extension.conf:
>
> exten => 4,1,Dial(IAX2/home_server:@${clinic_server}/${EXTEN},60,rw)
> exten => 4,n,Hangup() Remote Asterisk iax.conf:
>
> [home_server]
> type=friend
> host=dynamic
> secret=
> context=extensions
> disallow=all
> allow=ulaw
> allow=alaw
> callgroup=1
> pickupgroup=1
>
> Remote extension.conf:
>
> exten => 4,1,Dial(SIP/4,15,trw)
> exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
> exten => 4,n(line2),Dial(SIP/54,20,rw)
> exten => 4,n(vmail),Voicemail(4)
> exten => 4,n,Voicemail(4)
> exten => 4,n,Hangup()
>
>
You have no internal context in your dialplan. But in your iax.conf you specify
internal as your context.
-H
--
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Re: [asterisk-users] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
On 1/2/24 15:13, aster...@phreaknet.org wrote:
On 1/2/2024 4:03 PM, the...@sys-concept.com wrote:
I'm using asterisk-16.30.1
When I try to call another asterisk server over IAX I get a busy signal,
chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
-- IAX2/192.168.143.1:4569-656 is circuit-busy
Asterisk-16.16 is working normally, no congestion error.
There is not enough information for anyone to really help or comment on this.
Dialplan and IAX2 configuration on both sides of the trunk?
CLI output on both sides with iax2 debug enabled?
It is very simple:
Local Asterisk, iax.conf:
[clinic_server]
type=friend
host=dynamic
context=internal
disallow=all
allow=ulaw
allow=alaw
requirecalltoken=no
callgroup=1
pickupgroup=1
extension.conf:
exten => 4,1,Dial(IAX2/home_server:@${clinic_server}/${EXTEN},60,rw)
exten => 4,n,Hangup() Remote Asterisk iax.conf:
[home_server]
type=friend
host=dynamic
secret=
context=extensions
disallow=all
allow=ulaw
allow=alaw
callgroup=1
pickupgroup=1
Remote extension.conf:
exten => 4,1,Dial(SIP/4,15,trw)
exten => 4,n,GotoIf($["${DIALSTATUS}"="BUSY"]?vmail:line2)
exten => 4,n(line2),Dial(SIP/54,20,rw)
exten => 4,n(vmail),Voicemail(4)
exten => 4,n,Voicemail(4)
exten => 4,n,Hangup()
--
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Re: [asterisk-users] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
On 1/2/2024 4:03 PM, the...@sys-concept.com wrote: I'm using asterisk-16.30.1 When I try to call another asterisk server over IAX I get a busy signal, chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.143.1:4569-656 is circuit-busy Asterisk-16.16 is working normally, no congestion error. There is not enough information for anyone to really help or comment on this. Dialplan and IAX2 configuration on both sides of the trunk? CLI output on both sides with iax2 debug enabled? -- _ -- 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] chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response
I'm using asterisk-16.30.1 When I try to call another asterisk server over IAX I get a busy signal, chan_iax2.c:4739 __auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.143.1:4569-656 is circuit-busy Asterisk-16.16 is working normally, no congestion error. -- Thelma -- _ -- 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] Mailing List Future
On Wed, Dec 13, 2023 at 8:40 AM Joshua C. Colp wrote: > On Mon, Dec 4, 2023 at 8:54 AM Joshua C. Colp wrote: > >> On Mon, Dec 4, 2023 at 8:52 AM Antony Stone < >> antony.st...@asterisk.open.source.it> wrote: >> >>> On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote: >>> >>> > The mailing list will not receive emails from the forums. What I was >>> > referring to is that Discourse does provide the ability to receive >>> emails >>> > for posts or threads you're interested in, and you are able to respond >>> over >>> > email to them as well. >>> >>> I use this forum via its email interface, and I agree that it works. >>> The >>> biggest disadvantage I experience is that although you can _reply_ to a >>> thread >>> via email, you cannot create a new one; you have to use the web forum >>> interface for that. >>> >>> I don't know whether the forum software used here could be modified to >>> allow >>> that - I raised the same point on the FreeSwitch forum and an admin >>> quite >>> happily turned it on. Maybe that could be investigated here? >>> >> >> It actually is turned on for some of the categories, but as it's a hosted >> instance I am limited in the available plugins and modification that can be >> done to make this more clear. We can add documentation on the docs site for >> it, and see if we can do something else. >> > > To follow-up on this, I reached out to Discourse and they gave a > suggestion on how to make it more evident (though not as nice as I would > hope). I'll be experimenting with it in January, that is: making it more > clearer/evident that email exists. > Just a reminder all regarding the time frame on the asterisk-users list, and the move to Discourse[1]. In regards to starting threads using email I have gone through and set up email addresses for the various categories. The hard part is communicating this, and the options Discourse gave weren't exactly the best. For the first attempt I have done the following: 1. Added a menu item at the top for "Starting Threads Over Email" 2. Created a forum post[2] which documents the categories and their email address If there's any other suggestions on it feel free to raise it. Cheers, [1] https://community.asterisk.org/ [2] https://community.asterisk.org/t/starting-threads-over-email/100275 -- Joshua C. Colp Director of Engineering | Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] [Maybe OT]: SIP Provider
On Tue, 2023-11-07 at 08:42 +0100, Luca Bertoncello wrote: > The best will be a free service, but if not, I don't want to pay too > much... > As said: I need a SIP Provider to have an italian number (better if I > can choose the prefix) only to receive calls. > > Any suggestion? Assuming that DIDWW have presence in Italy (I would be surprised if not) they can do this sort of thing. Cheers, Kingsley. -- *Please note Barritel will be closed from 5pm Friday the 22nd of December 2023 until 9am on Tuesday 2nd January 2024* -- _ -- 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 release 21.0.2
The Asterisk Development Team would like to announce the release of asterisk-21.0.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You! Change Log for Release asterisk-21.0.2 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - res_rtp_asterisk: Fix regression issues with DTLS client check User Notes: Upgrade Notes: Closed Issues: - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr() -- _ -- 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 release 20.5.2
The Asterisk Development Team would like to announce the release of asterisk-20.5.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You! Change Log for Release asterisk-20.5.2 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - res_rtp_asterisk: Fix regression issues with DTLS client check User Notes: Upgrade Notes: Closed Issues: - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr() -- _ -- 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 release 18.20.2
The Asterisk Development Team would like to announce the release of asterisk-18.20.2. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.2 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You! Change Log for Release asterisk-18.20.2 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - res_rtp_asterisk: Fix regression issues with DTLS client check User Notes: Upgrade Notes: Closed Issues: - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr() -- _ -- 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 release certified-18.9-cert7
The Asterisk Development Team would like to announce the release of Certified asterisk-18.9-cert7. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert7 and https://downloads.asterisk.org/pub/telephony/certified-asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You! Change Log for Release asterisk-certified-18.9-cert7 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-certified-18.9-cert7.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/certified-18.9-cert6...certified-18.9-cert7) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-certified-18.9-cert7.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - res_rtp_asterisk: Fix regression issues with DTLS client check User Notes: Upgrade Notes: Closed Issues: - #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used - #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when it shouldn't - #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs created by ast_sockaddr_from_pj_sockaddr() -- _ -- 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] CORRECTED asterisk release 21.0.1
The earlier announcement should not have had any User or Upgrade notes. The Asterisk Development Team would like to announce security release Asterisk 21.0.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.0.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release: - [Path traversal via AMI GetConfig allows access to outside files]( https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f ) - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation]( https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq ) - [PJSIP logging allows attacker to inject fake Asterisk log entries ]( https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7 ) - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 'update']( https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh ) Change Log for Release asterisk-21.0.1 Links: - [Full ChangeLog]( https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md) - [GitHub Diff]( https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1) - [Tarball]( https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - res_pjsip_header_funcs: Duplicate new header value, don't copy. - res_pjsip: disable raw bad packet logging - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list - manager.c: Prevent path traversal with GetConfig. User Notes: Upgrade Notes: Closed Issues: None -- _ -- 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] CORRECTED asterisk release certified-18.9-cert6
The earlier release announcement should NOT have had any User or Upgrade notes. The Asterisk Development Team would like to announce security release Certified Asterisk 18.9-cert6. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert6 and https://downloads.asterisk.org/pub/telephony/certified-asterisk The following security advisories were resolved in this release: - [Path traversal via AMI GetConfig allows access to outside files]( https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f ) - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation]( https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq ) - [PJSIP logging allows attacker to inject fake Asterisk log entries ]( https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7 ) - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 'update']( https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh ) Change Log for Release asterisk-certified-18.9-cert6 Links: - [Full ChangeLog]( https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-certified-18.9-cert6.md) - [GitHub Diff]( https://github.com/asterisk/asterisk/compare/certified-18.9-cert5...certified-18.9-cert6) - [Tarball]( https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-certified-18.9-cert6.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - res_pjsip_header_funcs: Duplicate new header value, don't copy. - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list - manager.c: Prevent path traversal with GetConfig. - res_pjsip: disable raw bad packet logging User Notes: Upgrade Notes: Closed Issues: None -- _ -- 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 release certified-18.9-cert6
andwidth 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 release 21.0.1
ImportVar and SetAMAFlags applications have now been removed. Closed Issues: None -- _ -- 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 release 20.5.1
The Asterisk Development Team would like to announce security release Asterisk 20.5.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.5.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release: - [Path traversal via AMI GetConfig allows access to outside files](https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f) - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq) - [PJSIP logging allows attacker to inject fake Asterisk log entries ](https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7) - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh) Change Log for Release asterisk-20.5.1 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - res_pjsip_header_funcs: Duplicate new header value, don't copy. - res_pjsip: disable raw bad packet logging - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list - manager.c: Prevent path traversal with GetConfig. User Notes: Upgrade Notes: Closed Issues: None -- _ -- 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 release 18.20.1
The Asterisk Development Team would like to announce security release Asterisk 18.20.1. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.20.1 and https://downloads.asterisk.org/pub/telephony/asterisk The following security advisories were resolved in this release: - [Path traversal via AMI GetConfig allows access to outside files](https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f) - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq) - [PJSIP logging allows attacker to inject fake Asterisk log entries ](https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7) - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh) Change Log for Release asterisk-18.20.1 Links: - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) Summary: - res_pjsip_header_funcs: Duplicate new header value, don't copy. - res_pjsip: disable raw bad packet logging - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list - manager.c: Prevent path traversal with GetConfig. User Notes: Upgrade Notes: Closed Issues: None -- _ -- 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] Mailing List Future
On Mon, Dec 4, 2023 at 8:54 AM Joshua C. Colp wrote: > On Mon, Dec 4, 2023 at 8:52 AM Antony Stone < > antony.st...@asterisk.open.source.it> wrote: > >> On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote: >> >> > The mailing list will not receive emails from the forums. What I was >> > referring to is that Discourse does provide the ability to receive >> emails >> > for posts or threads you're interested in, and you are able to respond >> over >> > email to them as well. >> >> I use this forum via its email interface, and I agree that it works. The >> biggest disadvantage I experience is that although you can _reply_ to a >> thread >> via email, you cannot create a new one; you have to use the web forum >> interface for that. >> >> I don't know whether the forum software used here could be modified to >> allow >> that - I raised the same point on the FreeSwitch forum and an admin quite >> happily turned it on. Maybe that could be investigated here? >> > > It actually is turned on for some of the categories, but as it's a hosted > instance I am limited in the available plugins and modification that can be > done to make this more clear. We can add documentation on the docs site for > it, and see if we can do something else. > To follow-up on this, I reached out to Discourse and they gave a suggestion on how to make it more evident (though not as nice as I would hope). I'll be experimenting with it in January, that is: making it more clearer/evident that email exists. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] retry loop in ansible ?
I have a simple config with some phones ringing simultaneously. Some of them are softphones (zoiper apps on iPhone w/o push notification). If such an app did bot register in time, it has no chance to pick up the call. If I could configure a retry loop checking for registered candidates, say once a second until one phone takes the call, this would allow me to pick up the call with zoiper app registered late. How could this be done in ansible? Axel --- PGP-Key: CDE74120 ☀ mobile: +49 160 7568212 computing @ chaos claudius -- _ -- 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 13 / chan_sip / registration after reject
Hello > > How do I achieve the same with chan_sip? > We run a cron script each 10min who will check the registration state > and send a register if not registered. Well it's a simple CPE which needs to be autoprovisioned via either a tftp config file or TR69. So that cronjob somehow would also need to be put on the device via one of those mechanism. We check if there is a way. Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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] Mailing List Future
On Mon, Dec 4, 2023 at 8:52 AM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote: > > > The mailing list will not receive emails from the forums. What I was > > referring to is that Discourse does provide the ability to receive emails > > for posts or threads you're interested in, and you are able to respond > over > > email to them as well. > > I use this forum via its email interface, and I agree that it works. The > biggest disadvantage I experience is that although you can _reply_ to a > thread > via email, you cannot create a new one; you have to use the web forum > interface for that. > > I don't know whether the forum software used here could be modified to > allow > that - I raised the same point on the FreeSwitch forum and an admin quite > happily turned it on. Maybe that could be investigated here? > It actually is turned on for some of the categories, but as it's a hosted instance I am limited in the available plugins and modification that can be done to make this more clear. We can add documentation on the docs site for it, and see if we can do something else. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] Mailing List Future
On Monday 04 December 2023 at 13:39:51, Joshua C. Colp wrote: > The mailing list will not receive emails from the forums. What I was > referring to is that Discourse does provide the ability to receive emails > for posts or threads you're interested in, and you are able to respond over > email to them as well. I use this forum via its email interface, and I agree that it works. The biggest disadvantage I experience is that although you can _reply_ to a thread via email, you cannot create a new one; you have to use the web forum interface for that. I don't know whether the forum software used here could be modified to allow that - I raised the same point on the FreeSwitch forum and an admin quite happily turned it on. Maybe that could be investigated here? Antony. -- René Descartes walks in to a bar. The barman asks him "Do you want a drink?" Descartes says "I think not," and disappears. 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
Re: [asterisk-users] Mailing List Future
The mailing list will not receive emails from the forums. What I was referring to is that Discourse does provide the ability to receive emails for posts or threads you're interested in, and you are able to respond over email to them as well. On Mon, Dec 4, 2023 at 8:38 AM John Novack wrote: > > > Frank Vanoni wrote: > > On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > > > >> To that end, we’ve decided to discontinue the mailing lists effective > >> February 1st, 2024. > > That's a very sad news! :-( > > > Agree. Yet another giant step backward. > Interesting that they will continue to send e-mails when postings to the > (UGH) forum happen though. > > John Novack > > > > -- > Dog is my Co-Pilot > > > -- > _ > -- 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 -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] Mailing List Future
Frank Vanoni wrote: On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: To that end, we’ve decided to discontinue the mailing lists effective February 1st, 2024. That's a very sad news! :-( Agree. Yet another giant step backward. Interesting that they will continue to send e-mails when postings to the (UGH) forum happen though. John Novack -- Dog is my Co-Pilot -- _ -- 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] Mailing List Future
On Mon, 2023-12-04 at 08:00 -0400, Joshua C. Colp wrote: > To that end, we’ve decided to discontinue the mailing lists effective > February 1st, 2024. That's a very sad news! :-( -- _ -- 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] Mailing List Future
Greetings all, Over the past few years, the use of the Asterisk mailing lists has diminished, with far more conversation happening on the Asterisk community forums[1]. The state of email, to ensure reliable delivery, has also gotten more complicated - emails get caught by spam filters, etc.. To continue the mailing lists would require a huge time and resource investment, for minimal use. To that end, we’ve decided to discontinue the mailing lists effective February 1st, 2024. This means the following: 1. Sending and receiving mailing list emails will no longer be possible. 2. The list archives, however, will remain available. We recommend those who have not already done so migrate to the Asterisk Community forums[1]. You can choose to receive emails for posts if you wish, or purely use the web interface. You’re also able to privately message other individuals if you wish. Scoped categories also exist for more specific help. Cheers, [1] https://community.asterisk.org/ -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] Asterisk 13 / chan_sip / registration after reject
Hello Le 04/12/2023 à 10:56, Benoit Panizzon (by way of Benoit Panizzon ) a écrit : Hi List We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure: auth_rejection_permanent=no How do I achieve the same with chan_sip? We run a cron script each 10min who will check the registration state and send a register if not registered. -- 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
[asterisk-users] Asterisk 13 / chan_sip / registration after reject
Hi List We have some CPE which run an embedded asterisk 13 with chan_sip. Unfortunately, when a registration is rejected, those stop trying. I am familiar with pjsip which allows to configure: auth_rejection_permanent=no How do I achieve the same with chan_sip? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- 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] Recommended sip providers
Please contact billing at chaneste dot com Route is flat fee 0.0065 with Stir Shaken included. Nine_ five_ 4 triple 4 se_ven four_ ze_ro _eight From: asterisk-users On Behalf Of Tahir Almas Dhesi Sent: Monday, November 20, 2023 6:14 AM To: Commercial and Business-Oriented Asterisk Discussion ; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Recommended sip providers Interested to know a good wholesale sip providers for 15k concurrent calls regards Tahir Almas Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- 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] help with crash
] asterisk pbx.c:4377 __ast_pbx_run() #18: [0x53184b] asterisk pbx.c:4669 decrease_call_count() #19: [inlined] asterisk pbx.c:4702 pbx_thread() #20: [0x5b8329] asterisk utils.c:1576 dummy_start() #21: [0x7f62bec07ea5] libpthread.so.0 :0 __pthread_get_minstack() #22: [0x7f62bd0fe8dd] libc.so.6 :0 clone() [2023-11-08 18:14:14] ERROR[571354][C-17e4] stasis_cache.c: Excessive refcount 10 reached on ao2 object 0x3616b38 [2023-11-08 18:14:14] ERROR[571354][C-17e4] stasis_cache.c: FRACK!, Failed assertion Excessive refcount 10 reached on ao2 object 0x3616b38 (0) [2023-11-08 18:14:15] ERROR[571354][C-17e4] : Got 23 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() -- _ -- 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] Finding old patches
On Mon, Nov 20, 2023 at 1:45 PM Dovid Bender wrote: > Hi, > > In the past when I wanted to back port a patch I would go on to the issue > tracker and find a link to the patches that were uploaded ( I think > through gerrit?). I am trying to see what changes were done for > https://issues-archive.asterisk.org/ASTERISK-26109. It seems the code > changes were introduced in 14.4.0-rc1. Is there any "n00b" way of seeing > what patches were created for this specific issue? > The git commit history can be used, the commit message can be grepped, example: ✔ jcolp@kappa:~/development/asterisk/github [21| …2⚑ 3]> git log | grep -B4 "OpenSSL 1.1.0 support" Merge: a0c0b1c9cb 26c8552fff Author: zuul Date: Wed Nov 30 23:26:46 2016 -0600 Merge "OpenSSL 1.1.0 support" -- commit 26c8552fff499419bdf12b663e76ecfc408b3085 Author: Tzafrir Cohen Date: Tue Jun 28 23:26:59 2016 +0200 OpenSSL 1.1.0 support So the commit is "26c8552fff499419bdf12b663e76ecfc408b3085" and you can use git show to display that commit, which includes the changes including in diff format: ✔ jcolp@kappa:~/development/asterisk/github [21| …2⚑ 3]> git show 26c8552fff commit 26c8552fff499419bdf12b663e76ecfc408b3085 Author: Tzafrir Cohen Date: Tue Jun 28 23:26:59 2016 +0200 OpenSSL 1.1.0 support OpenSSL 1.1.0 includes some major changes in the interface. See https://wiki.openssl.org/index.php/1.1_API_Changes . Status: Right now there are still a few deprecation notes with OpenSSL 1.1.0. But it's a start. Changes: * CRYPTO_LOCK is no longer available. Replace it with its value for now. I don't completely understand what it is used for there. * Remove several functions from libasteriskssl that seem to no longer be needed. * Structures have become opaque and are accesses with accessors. * ERR_remove_thread_state() no longer needed. * SSLv2 code now could no longer be used in 1.1. ASTERISK-26109 #close Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b diff --git a/main/iostream.c b/main/iostream.c The same goes for the rest of the associated commits on the linked issue. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] Finding old patches
Hi, In the past when I wanted to back port a patch I would go on to the issue tracker and find a link to the patches that were uploaded ( I think through gerrit?). I am trying to see what changes were done for https://issues-archive.asterisk.org/ASTERISK-26109. It seems the code changes were introduced in 14.4.0-rc1. Is there any "n00b" way of seeing what patches were created for this specific issue? TIA. Dovid -- _ -- 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] Recommended sip providers
Only outbound to USA so no DID Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT On Mon, Nov 20, 2023 at 4:18 PM Antony Stone < antony.st...@asterisk.open.source.it> wrote: > On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote: > > > Interested to know good wholesale SIP providers for 15k concurrent calls > > You might want to specify a bit more detail, such as: > > - which country are you located in > - do you require inbound DDIs (if so, in which region/s)? > - which countries' Caller ID/s do you need to present? > > Antony. > > -- > These clients are often infected by viruses or other malware and need to > be > fixed. If not, the user at that client needs to be fixed... > > - Henrik Nordstrom, on Squid users' mailing list > >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 -- _ -- 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] Recommended sip providers
On Monday 20 November 2023 at 12:14:11, Tahir Almas Dhesi wrote: > Interested to know good wholesale SIP providers for 15k concurrent calls You might want to specify a bit more detail, such as: - which country are you located in - do you require inbound DDIs (if so, in which region/s)? - which countries' Caller ID/s do you need to present? Antony. -- These clients are often infected by viruses or other malware and need to be fixed. If not, the user at that client needs to be fixed... - Henrik Nordstrom, on Squid users' mailing list 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] Recommended sip providers
Interested to know a good wholesale sip providers for 15k concurrent calls regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT -- _ -- 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] [Maybe OT]: SIP Provider
Hi On 07/11/2023 08:42, Luca Bertoncello wrote: Currently I'm using Messagenet, a SIP-Provider in Italy, to have an italian number via VoIP, _to receive calls only_. I use it to allow my friends and parents in Italy to call me in Germany without paying too much. This service was free of charge in the last years. Now will Messagenet beginning from end of november, to cancel this free service and only offer paying services (for receiving and calling). Since I don't need to call from this number (using Deutsche Telekom I already can call Italy for free), I'm currently searching an alternative. The best will be a free service, but if not, I don't want to pay too much... As said: I need a SIP Provider to have an italian number (better if I can choose the prefix) only to receive calls. Any suggestion? An overview; https://www.voipkredi.com/page.php?page=betamax-dellmont There are probably other pages like this. Regards, Rob -- _ -- 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] Enterqueue event not generated when cfu internal
Hi all Using asterisk 16.25.0 (I know it's a bit old) I'm trying to parse the realtime queue_log and I realized that not every call has a ENTERQUEUE event. I see that if there's an internal call to an extension that has a dialplan forward to a queue (no call is placed to the extension before calling the queue), the ENTERQUEUE event is not generated in the queue_log For calls from the pstn the event exists. Both type of calls (incoming and internal) call the same Gosub to call the queue. I don't get it why the events should be different. any hints? thank you. -- PekePBX, the multitenant PBX solution https://pekepbx.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
[asterisk-users] help with crash
23 backtrace records # 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed() -- _ -- 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] SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED
= sha-256 Why is there "UNSUPPORTED OR FAILED" in the log when processing "a=ice-ufrag" and "ice-pwd" ?? Asterisk gives no "a=ice-ufrag" and "ice-pwd" in its "SIP/2.0 200 OK" response to the INVITE and thus sipjs aborts the SIP call with a 488-"Not Acceptable Here". I read about libuuid-devel and uuid-devel are necessary, but I have these installed ! What am I missing here ?! Kind regards. -- _ -- 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] Local calls not possible when Internet connection down
Hello, I confirm server and phones are on the same subnet and the phones are able to resolve local domain also when internet connection os down. It seems to be the asterisk bug I referenced before. There seems to be some bolcking resolver in it. I do not use database related to asterisk. This should be related to the srv record resolving. It seems quite random time to trigger the issue. When inspecting logs after internet problems started the issue appeared in one hour and several minutes. After restart of the asterisk it reappeared in less than half an hour. When trying to reproduce I was not able to reproduce for one hour and a half. So I decided to configure srv_lookups=no. I hope the issue is workarounded now. But I think asterisk should be fixed. It should successfully start when the VoIP providers sip server is not reachable, should recover after it becomes available. And should work locally when it stops to be responding. The tweak of creating /etc/hosts entry for the sip server and disabling srv lookups should not be needed. I hope sometimes theese issues will be addressed. Marek On Wednesday, November 8th, 2023 at 15:53, John Harragin wrote: > Are the phones and the server in the same subnet? You might making note of > the IPs and just simply try pinging everything with the uplink disconnected. > Also, if you are using domain names for registration, it is possible a dns > server must be reachable. > > If you are using database for any of your call processing, an unreachable dns > server can also be the cause of trouble. For some reason, even if you are > using IP addressing, Mysql will try to resolve a connection and can hang > (there is a mysql parameter to not resolve addresses). > > On Wed, Nov 8, 2023 at 8:46 AM Marek Greško > wrote: > >> Hello, >> >> it did not seem the call hung. It seemed it never started. There was no >> dialplan execution on the asterisk side. It looked like phones were >> unregistered. Same shows the log posted previously. >> >> Marek >> >> Sent with Proton Mail secure email. >> >> --- Original Message --- >> On Wednesday, November 8th, 2023 at 1:21, John Harragin >> wrote: >> >>> Marek, >>> >>> See if calls hang in the system if you encounter another outage >>> core show channels >>> >>> ...if so, >>> core set verbose 3 >>> and see what instructions subsequent calls hang on. >>> >>> >>> >>> On Mon, Nov 6, 2023 at 4:44 PM Marek Greško marek.gre...@protonmail.com >>> wrote: >>> >>> > Hello, >>> > >>> > sure I have local DNS server and public resolving should not be needed >>> > for phone registrations. Running pjsip show endpojnt show the endpoints >>> > as not in use. >>> > >>> > When looking into logs I see only res_pjsip_outbound_registration.c: No >>> > response >>> > received from sip provider. Nothing else. >>> > >>> > In phone log I see: >>> > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), >>> > lid=0, par=0, par2=(nil)) >>> > >>> > The phone is Cisco SPA525G2. >>> > >>> > Thanks. >>> > >>> > Marek >>> > >>> > --- Original Message --- >>> > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp jc...@sangoma.com >>> > wrote: >>> > >>> > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško marek.gre...@protonmail.com >>> > wrote: >>> > >>> > > It looks like all phones get unregistered, but I am not aware of the >>> > > cause. Why are get not registered when there is a connectivity between >>> > > them and asterisk? >>> > >>> > Are the REGISTER requests reaching Asterisk (do they show up in a packet >>> > capture, do they show up in "pjsip set logger on")? It needs to be >>> > further isolated. How are the phones configured to reach Asterisk? If >>> > using a hostname, are they still able to resolve it? >>> > >>> > -- >>> > Joshua C. Colp >>> > Asterisk Project Lead >>> > Sangoma Technologies >>> > Check us out at www.sangoma.com and www.asterisk.org >>> > >>> > -- >>> > _ >>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> > >>> > Ch
Re: [asterisk-users] Local calls not possible when Internet connection down
Are the phones and the server in the same subnet? You might making note of the IPs and just simply try pinging everything with the uplink disconnected. Also, if you are using domain names for registration, it is possible a dns server must be reachable. If you are using database for any of your call processing, an unreachable dns server can also be the cause of trouble. For some reason, even if you are using IP addressing, Mysql will try to resolve a connection and can hang (there is a mysql parameter to not resolve addresses). On Wed, Nov 8, 2023 at 8:46 AM Marek Greško wrote: > Hello, > > it did not seem the call hung. It seemed it never started. There was no > dialplan execution on the asterisk side. It looked like phones were > unregistered. Same shows the log posted previously. > > Marek > > > > > > Sent with Proton Mail secure email. > > --- Original Message --- > On Wednesday, November 8th, 2023 at 1:21, John Harragin < > jharra...@mw.k12.ny.us> wrote: > > > > Marek, > > > > See if calls hang in the system if you encounter another outage > > core show channels > > > > ...if so, > > core set verbose 3 > > and see what instructions subsequent calls hang on. > > > > > > > > On Mon, Nov 6, 2023 at 4:44 PM Marek Greško marek.gre...@protonmail.com > wrote: > > > > > Hello, > > > > > > sure I have local DNS server and public resolving should not be needed > for phone registrations. Running pjsip show endpojnt show the endpoints as > not in use. > > > > > > When looking into logs I see only res_pjsip_outbound_registration.c: > No response > > > received from sip provider. Nothing else. > > > > > > In phone log I see: > > > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > > > lid=0, par=0, par2=(nil)) > > > > > > The phone is Cisco SPA525G2. > > > > > > Thanks. > > > > > > Marek > > > > > > --- Original Message --- > > > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp > jc...@sangoma.com wrote: > > > > > > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško > marek.gre...@protonmail.com wrote: > > > > > > > It looks like all phones get unregistered, but I am not aware of the > cause. Why are get not registered when there is a connectivity between them > and asterisk? > > > > > > Are the REGISTER requests reaching Asterisk (do they show up in a > packet capture, do they show up in "pjsip set logger on")? It needs to be > further isolated. How are the phones configured to reach Asterisk? If using > a hostname, are they still able to resolve it? > > > > > > -- > > > Joshua C. Colp > > > Asterisk Project Lead > > > Sangoma Technologies > > > Check us out at www.sangoma.com and 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 > > > > > > -- > > _ > > -- 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 > > -- > _ > -- 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 -- _ -- 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] Local calls not possible when Internet connection down
Hello, it did not seem the call hung. It seemed it never started. There was no dialplan execution on the asterisk side. It looked like phones were unregistered. Same shows the log posted previously. Marek Sent with Proton Mail secure email. --- Original Message --- On Wednesday, November 8th, 2023 at 1:21, John Harragin wrote: > Marek, > > See if calls hang in the system if you encounter another outage > core show channels > > ...if so, > core set verbose 3 > and see what instructions subsequent calls hang on. > > > > On Mon, Nov 6, 2023 at 4:44 PM Marek Greško marek.gre...@protonmail.com wrote: > > > Hello, > > > > sure I have local DNS server and public resolving should not be needed for > > phone registrations. Running pjsip show endpojnt show the endpoints as not > > in use. > > > > When looking into logs I see only res_pjsip_outbound_registration.c: No > > response > > received from sip provider. Nothing else. > > > > In phone log I see: > > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > > lid=0, par=0, par2=(nil)) > > > > The phone is Cisco SPA525G2. > > > > Thanks. > > > > Marek > > > > --- Original Message --- > > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp jc...@sangoma.com > > wrote: > > > > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško marek.gre...@protonmail.com > > wrote: > > > > > It looks like all phones get unregistered, but I am not aware of the > > > cause. Why are get not registered when there is a connectivity between > > > them and asterisk? > > > > Are the REGISTER requests reaching Asterisk (do they show up in a packet > > capture, do they show up in "pjsip set logger on")? It needs to be further > > isolated. How are the phones configured to reach Asterisk? If using a > > hostname, are they still able to resolve it? > > > > -- > > Joshua C. Colp > > Asterisk Project Lead > > Sangoma Technologies > > Check us out at www.sangoma.com and 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 > > > -- > _____ > -- 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 -- _____ -- 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] Local calls not possible when Internet connection down
Marek, See if calls hang in the system if you encounter another outage core show channels ...if so, core set verbose 3 and see what instructions subsequent calls hang on. On Mon, Nov 6, 2023 at 4:44 PM Marek Greško wrote: > > Hello, > > sure I have local DNS server and public resolving should not be needed for > phone registrations. Running pjsip show endpojnt show the endpoints as not in > use. > > When looking into logs I see only res_pjsip_outbound_registration.c: No > response > received from sip provider. Nothing else. > > In phone log I see: > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > lid=0, par=0, par2=(nil)) > > The phone is Cisco SPA525G2. > > Thanks. > > Marek > > > > --- Original Message --- > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp > wrote: > > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško > wrote: >> >> It looks like all phones get unregistered, but I am not aware of the cause. >> Why are get not registered when there is a connectivity between them and >> asterisk? > > > Are the REGISTER requests reaching Asterisk (do they show up in a packet > capture, do they show up in "pjsip set logger on")? It needs to be further > isolated. How are the phones configured to reach Asterisk? If using a > hostname, are they still able to resolve it? > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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 -- _ -- 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] Local calls not possible when Internet connection down
Hello Joshua, thanks for suggestion. I just found out the same solution several minutes ago. I also obtained the maintenance window, so I diasbled outgoing DNS and SIP. But I was not successful reproducing the bad state. So I ceased futher debugging attempts and set srv_lookups to no. We will see once the next massive internet outage out of my control happens, whether it helped. Thanks again Marek --- Original Message --- On Tuesday, November 7th, 2023 at 16:28, Joshua C. Colp wrote: > On Tue, Nov 7, 2023 at 11:20 AM Marek Greško > wrote: > >> Hello, >> >> well I do not ask those who only guess, but those who know what is asterisk >> expected to do when internet connectivity goes down. I did not had a chance >> to make internet not to work yet, since it is needed. But inspecting dns >> logs I found out that there started to be resolving for _sip._tcp and >> _sip._udp records for the provider's server. So apparently making hosts >> record make asterisk happy when everything works, but when there is a >> communication problem then it falls back to searching for srv records. At >> least it seems to be so for now. Moreover I found out this old thread: > > The expectation is that Asterisk continues to work. That being said there is > one case (specifically using realtime with an identify section that > references a hostname) that can cause this specific behavior where PJSIP will > block. > > Are you in that scenario? If so you CAN disable SRV records on the identify > by setting "srv_lookups" to "no". > -- > > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
On Tue, Nov 7, 2023 at 11:20 AM Marek Greško wrote: > Hello, > > well I do not ask those who only guess, but those who know what is > asterisk expected to do when internet connectivity goes down. I did not had > a chance to make internet not to work yet, since it is needed. But > inspecting dns logs I found out that there started to be resolving for > _sip._tcp and _sip._udp records for the provider's server. So apparently > making hosts record make asterisk happy when everything works, but when > there is a communication problem then it falls back to searching for srv > records. At least it seems to be so for now. Moreover I found out this old > thread: > The expectation is that Asterisk continues to work. That being said there is one case (specifically using realtime with an identify section that references a hostname) that can cause this specific behavior where PJSIP will block. Are you in that scenario? If so you CAN disable SRV records on the identify by setting "srv_lookups" to "no". -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
Hello, well I do not ask those who only guess, but those who know what is asterisk expected to do when internet connectivity goes down. I did not had a chance to make internet not to work yet, since it is needed. But inspecting dns logs I found out that there started to be resolving for _sip._tcp and _sip._udp records for the provider's server. So apparently making hosts record make asterisk happy when everything works, but when there is a communication problem then it falls back to searching for srv records. At least it seems to be so for now. Moreover I found out this old thread: https://community.freepbx.org/t/asterisk-become-mad-when-a-dns-problem-occur/4755/10 So the problem seems to be still present. So if asterisk is not able to resolve using it's dns resolver it breaks also local communication which is complete non-sense. I am thinking of two possible workarounds: 1. If thre is a possibility to convince asterisk not to fallback to searching for srv records, it would be ideal. Is somebody aware of such options in pjsip? 2. If the first workaround is not feasible I can create rpz records for provider's A and SRV records. When I will be able to shutdown internet or at least outbound DNS, I will try to make sure my findings are correct using tcpdump. Thanks Marek Sent with Proton Mail secure email. --- Original Message --- On Tuesday, November 7th, 2023 at 0:46, Greg Troxel wrote: > Marek Greško marek.gre...@protonmail.com writes: > > > But I am not sure why this is happening. I have sip providers hostname > > in /etc/hosts file to prevent such situations. Should I reconfigure it > > not to use hosts file but rather some RPZ on DNS server? Does asterisk > > ignore hosts file? Or does it try to do some srv lookups? But in > > either case, why does this influence local calls? Local domain should > > really be resolvable. > > > You should run tcpdump on 53 and 5353 in multiple places and figure out > what it is doing, rather than asking us, who can only guess. -- _ -- 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] [Maybe OT]: SIP Provider
Hi all! Currently I'm using Messagenet, a SIP-Provider in Italy, to have an italian number via VoIP, _to receive calls only_. I use it to allow my friends and parents in Italy to call me in Germany without paying too much. This service was free of charge in the last years. Now will Messagenet beginning from end of november, to cancel this free service and only offer paying services (for receiving and calling). Since I don't need to call from this number (using Deutsche Telekom I already can call Italy for free), I'm currently searching an alternative. The best will be a free service, but if not, I don't want to pay too much... As said: I need a SIP Provider to have an italian number (better if I can choose the prefix) only to receive calls. Any suggestion? Thanks a lot 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] Local calls not possible when Internet connection down
Hello, the corresponding conf is: pbx.example.lan No Yes Yes No 3600 No No No 3600 Normal No No Marek --- Original Message --- On Tuesday, November 7th, 2023 at 0:22, Łukasz Grzywański wrote: > Could you show the phone configurations - section "Proxy and Registration" > > On Mon, 6 Nov 2023 at 23:13, Marek Greško wrote: > >> Hello, >> >> you are probably right. It should somehow be related to DNS. I just found >> out this in the storm of previous messages: >> >> WARNING[13945] taskprocessor.c: The 'dns_system_resolver_tp' task processor >> queue reached 500 scheduled tasks. >> >> But I am not sure why this is happening. I have sip providers hostname in >> /etc/hosts file to prevent such situations. Should I reconfigure it not to >> use hosts file but rather some RPZ on DNS server? Does asterisk ignore hosts >> file? Or does it try to do some srv lookups? But in either case, why does >> this influence local calls? Local domain should really be resolvable. >> >> Thanks >> >> Marek >> >> --- Original Message --- >> On Monday, November 6th, 2023 at 19:52, Marek Greško >> wrote: >> >>> Hello, >>> >>> sure I have local DNS server and public resolving should not be needed for >>> phone registrations. Running pjsip show endpojnt show the endpoints as not >>> in use. >>> >>> When looking into logs I see only res_pjsip_outbound_registration.c: No >>> response >>> received from sip provider. Nothing else. >>> >>> In phone log I see: >>> CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), >>> lid=0, par=0, par2=(nil)) >>> >>> The phone is Cisco SPA525G2. >>> >>> Thanks. >>> >>> Marek >>> >>> --- Original Message --- >>> On Monday, November 6th, 2023 at 15:45, Joshua C. Colp >>> wrote: >>> >>>> On Mon, Nov 6, 2023 at 10:42 AM Marek Greško >>>> wrote: >>>> >>>>> It looks like all phones get unregistered, but I am not aware of the >>>>> cause. Why are get not registered when there is a connectivity between >>>>> them and asterisk? >>>> >>>> Are the REGISTER requests reaching Asterisk (do they show up in a packet >>>> capture, do they show up in "pjsip set logger on")? It needs to be further >>>> isolated. How are the phones configured to reach Asterisk? If using a >>>> hostname, are they still able to resolve it? >>>> -- >>>> >>>> Joshua C. Colp >>>> Asterisk Project Lead >>>> Sangoma Technologies >>>> Check us out at www.sangoma.com and 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 > > -- > > Pozdrawiam, > > Łukasz Grzywański > Voice Architect > > Mok Yok IT Sp. z o.o. > ul. Rzeźnicza 32/33, 50-130 Wrocław, Polska > tel. +48 717227200, fax +48 717227299 > mob.: +48 517255333, e-mail: lukasz.grzywan...@mokyokit.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] Local calls not possible when Internet connection down
Marek Greško writes: > But I am not sure why this is happening. I have sip providers hostname > in /etc/hosts file to prevent such situations. Should I reconfigure it > not to use hosts file but rather some RPZ on DNS server? Does asterisk > ignore hosts file? Or does it try to do some srv lookups? But in > either case, why does this influence local calls? Local domain should > really be resolvable. You should run tcpdump on 53 and 5353 in multiple places and figure out what it is doing, rather than asking us, who can only guess. -- _ -- 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] Local calls not possible when Internet connection down
Could you show the phone configurations - section "Proxy and Registration" On Mon, 6 Nov 2023 at 23:13, Marek Greško wrote: > Hello, > > you are probably right. It should somehow be related to DNS. I just found > out this in the storm of previous messages: > > WARNING[13945] taskprocessor.c: The 'dns_system_resolver_tp' task > processor queue reached 500 scheduled tasks. > > But I am not sure why this is happening. I have sip providers hostname in > /etc/hosts file to prevent such situations. Should I reconfigure it not to > use hosts file but rather some RPZ on DNS server? Does asterisk ignore > hosts file? Or does it try to do some srv lookups? But in either case, why > does this influence local calls? Local domain should really be resolvable. > > Thanks > > Marek > > > --- Original Message --- > On Monday, November 6th, 2023 at 19:52, Marek Greško < > marek.gre...@protonmail.com> wrote: > > Hello, > > sure I have local DNS server and public resolving should not be needed for > phone registrations. Running pjsip show endpojnt show the endpoints as not > in use. > > When looking into logs I see only res_pjsip_outbound_registration.c: No > response > received from sip provider. Nothing else. > > In phone log I see: > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > lid=0, par=0, par2=(nil)) > > The phone is Cisco SPA525G2. > > Thanks. > > Marek > > > > --- Original Message --- > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp > wrote: > > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško > wrote: > >> It looks like all phones get unregistered, but I am not aware of the >> cause. Why are get not registered when there is a connectivity between them >> and asterisk? >> > > Are the REGISTER requests reaching Asterisk (do they show up in a packet > capture, do they show up in "pjsip set logger on")? It needs to be further > isolated. How are the phones configured to reach Asterisk? If using a > hostname, are they still able to resolve it? > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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 -- Pozdrawiam, Łukasz Grzywański Voice Architect Mok Yok IT Sp. z o.o. ul. Rzeźnicza 32/33, 50-130 Wrocław, Polska tel. +48 717227200, fax +48 717227299 mob.: +48 517255333, e-mail: lukasz.grzywan...@mokyokit.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] Local calls not possible when Internet connection down
Hello, you are probably right. It should somehow be related to DNS. I just found out this in the storm of previous messages: WARNING[13945] taskprocessor.c: The 'dns_system_resolver_tp' task processor queue reached 500 scheduled tasks. But I am not sure why this is happening. I have sip providers hostname in /etc/hosts file to prevent such situations. Should I reconfigure it not to use hosts file but rather some RPZ on DNS server? Does asterisk ignore hosts file? Or does it try to do some srv lookups? But in either case, why does this influence local calls? Local domain should really be resolvable. Thanks Marek --- Original Message --- On Monday, November 6th, 2023 at 19:52, Marek Greško wrote: > Hello, > > sure I have local DNS server and public resolving should not be needed for > phone registrations. Running pjsip show endpojnt show the endpoints as not in > use. > > When looking into logs I see only res_pjsip_outbound_registration.c: No > response > received from sip provider. Nothing else. > > In phone log I see: > CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), > lid=0, par=0, par2=(nil)) > > The phone is Cisco SPA525G2. > > Thanks. > > Marek > > --- Original Message --- > On Monday, November 6th, 2023 at 15:45, Joshua C. Colp > wrote: > >> On Mon, Nov 6, 2023 at 10:42 AM Marek Greško >> wrote: >> >>> It looks like all phones get unregistered, but I am not aware of the cause. >>> Why are get not registered when there is a connectivity between them and >>> asterisk? >> >> Are the REGISTER requests reaching Asterisk (do they show up in a packet >> capture, do they show up in "pjsip set logger on")? It needs to be further >> isolated. How are the phones configured to reach Asterisk? If using a >> hostname, are they still able to resolve it? >> -- >> >> Joshua C. Colp >> Asterisk Project Lead >> Sangoma Technologies >> Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
Hello, sure I have local DNS server and public resolving should not be needed for phone registrations. Running pjsip show endpojnt show the endpoints as not in use. When looking into logs I see only res_pjsip_outbound_registration.c: No response received from sip provider. Nothing else. In phone log I see: CC_eventProc(event=63(CC_EV_SIG_REGISTER_FAILED), lid=0, par=0, par2=(nil)) The phone is Cisco SPA525G2. Thanks. Marek --- Original Message --- On Monday, November 6th, 2023 at 15:45, Joshua C. Colp wrote: > On Mon, Nov 6, 2023 at 10:42 AM Marek Greško > wrote: > >> It looks like all phones get unregistered, but I am not aware of the cause. >> Why are get not registered when there is a connectivity between them and >> asterisk? > > Are the REGISTER requests reaching Asterisk (do they show up in a packet > capture, do they show up in "pjsip set logger on")? It needs to be further > isolated. How are the phones configured to reach Asterisk? If using a > hostname, are they still able to resolve it? > -- > > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
Łukasz Grzywański writes: > I think it's a problem with DNS server availability I have tried to and mostly succeeded at making things work when the WAN is down. Elements needed: run a local named, vs configuring resolver to your ISP for names needed in the LAN, ensure they are answered locally without needing any non-local DNS records. For me this is "foo.local" being in a policy zone to provide the LAN address of the host, as the WAN address goes away and changes Consider if you want to just put LAN IP addresses in your config actually turn off the WAN and test when that works, pull the cable to the ONT from the router, or equivalent, to simulate failure vs just deconniguring it and test again -- _ -- 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] Local calls not possible when Internet connection down
On Mon, Nov 6, 2023 at 10:42 AM Marek Greško wrote: > It looks like all phones get unregistered, but I am not aware of the > cause. Why are get not registered when there is a connectivity between them > and asterisk? > Are the REGISTER requests reaching Asterisk (do they show up in a packet capture, do they show up in "pjsip set logger on")? It needs to be further isolated. How are the phones configured to reach Asterisk? If using a hostname, are they still able to resolve it? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
Hi Marek ! pls show logs :) I think it's a problem with DNS server availability Lukasz On Mon, 6 Nov 2023 at 15:41, Marek Greško wrote: > It looks like all phones get unregistered, but I am not aware of the > cause. Why are get not registered when there is a connectivity between them > and asterisk? > > Marek > > --- Original Message --- > On Monday, November 6th, 2023 at 15:10, Joshua C. Colp > wrote: > > On Mon, Nov 6, 2023 at 10:06 AM Marek Greško > wrote: > >> Hello, >> >> I just realized that when my Internet connection goes down and I loose >> connectivity to VoIP SIP provider I loose ability to make local calls after >> some time. When I restart asterisk, I am able to make local calls for some >> time, but it then suddenly stops working again. I am using pjsip stack. >> >> What could be the cause of this? >> > > There is insufficient information to be able to answer this. Such as, what > actually happens when attempts are made? What shows on the console? > > -- > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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 -- _ -- 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] Local calls not possible when Internet connection down
It looks like all phones get unregistered, but I am not aware of the cause. Why are get not registered when there is a connectivity between them and asterisk? Marek --- Original Message --- On Monday, November 6th, 2023 at 15:10, Joshua C. Colp wrote: > On Mon, Nov 6, 2023 at 10:06 AM Marek Greško > wrote: > >> Hello, >> >> I just realized that when my Internet connection goes down and I loose >> connectivity to VoIP SIP provider I loose ability to make local calls after >> some time. When I restart asterisk, I am able to make local calls for some >> time, but it then suddenly stops working again. I am using pjsip stack. >> >> What could be the cause of this? > > There is insufficient information to be able to answer this. Such as, what > actually happens when attempts are made? What shows on the console? > -- > > Joshua C. Colp > Asterisk Project Lead > Sangoma Technologies > Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
On Mon, Nov 6, 2023 at 10:06 AM Marek Greško wrote: > Hello, > > I just realized that when my Internet connection goes down and I loose > connectivity to VoIP SIP provider I loose ability to make local calls after > some time. When I restart asterisk, I am able to make local calls for some > time, but it then suddenly stops working again. I am using pjsip stack. > > What could be the cause of this? > There is insufficient information to be able to answer this. Such as, what actually happens when attempts are made? What shows on the console? -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com and 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] Local calls not possible when Internet connection down
Hello, I just realized that when my Internet connection goes down and I loose connectivity to VoIP SIP provider I loose ability to make local calls after some time. When I restart asterisk, I am able to make local calls for some time, but it then suddenly stops working again. I am using pjsip stack. What could be the cause of this? Thnaks Marek-- _ -- 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 and Teams integration?
On Thursday 26 October 2023 at 19:11:45, Carlos Chavez wrote: > Does anyone know of a good solution to integrate Asterisk and MS > Teams? Something where you can use the MS Teams client as a regular > extension? Kamailio is the usual intermediary I have seen for doing this. Antony. -- If at first you don't succeed, destroy all the evidence that you tried. 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 and Teams integration?
Does anyone know of a good solution to integrate Asterisk and MS Teams? Something where you can use the MS Teams client as a regular extension? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- 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 release 21.0.0
This also removes the 'w' and 'W' options for app_queue. MixMonitor should be default and only option for all settings that previously used either Monitor or MixMonitor. - ### app_osplookup: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### app_cdr: Remove deprecated application and option. The previously deprecated NoCDR application has been removed. Additionally, the previously deprecated 'e' option to the ResetCDR application has been removed. - ### app_macro: Remove deprecated module. This module was deprecated in Asterisk 16 and is now being removed in accordance with the Asterisk Module Deprecation policy. For most modules that interacted with app_macro, this change is limited to no longer looking for the current context from the macrocontext when set. The following modules have additional impacts: app_dial - no longer supports M^ connected/redirecting macro app_minivm - samples written using macro will no longer work. The sample needs to be re-written app_queue - can no longer call a macro on the called party's channel. Use gosub which is currently supported ccss - no callback macro, gosub only app_voicemail - no macro support channel - remove macrocontext and priority, no connected line or redirection macro options options - stdexten is deprecated to gosub as the default and only options pbx - removed macrolock pbx_dundi - no longer look for macro snmp - removed macro context, exten, and priority - ### translate.c: Prefer better codecs upon translate ties. When setting up translation between two codecs the quality was not taken into account, resulting in suboptimal translation. The quality is now taken into account, which can reduce the number of translation steps required, and improve the resulting quality. - ### chan_sip: Remove deprecated module. This module was deprecated in Asterisk 17 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### chan_alsa: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### pbx_builtins: Remove deprecated and defunct functionality. The previously deprecated ImportVar and SetAMAFlags applications have now been removed. - ### chan_mgcp: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. - ### chan_skinny: Remove deprecated module. This module was deprecated in Asterisk 19 and is now being removed in accordance with the Asterisk Module Deprecation policy. Closed Issues: - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms - #39: [Bug]: Remove .gitreview from repository. - #41: [Bug]: say.c Time announcement does not say o'clock for the French language - #50: [improvement]: app_sla: Migrate SLA applications from app_meetme - #78: [improvement]: Deprecate ast_gethostbyname() - #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime - #183: [deprecation]: Deprecate users.conf - #226: [improvement]: Apply contact_user to incoming calls - #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading - #240: [new-feature]: sig_analog: Add Called Subscriber Held capability - #253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs - #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered" - #263: [bug]: download_externals doesn't always handle versions correctly - #267: [bug]: ari: refer with display_name key in request body leads to crash - #274: [bug]: Syntax Error in SQL Code - #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement' - #277: [bug]: pbx.c: Compiler error with gcc 12.2 - #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits - #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in build_resource_tree -- _ -- 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 release 20.5.0
iority change immediately. The 'queue priority caller' CLI command and 'QueueChangePriorityCaller' AMI action now have an 'immediate' argument which allows the caller priority change to be reflected immediately, causing the position of a caller to move within the queue depending on the priorities of the other callers. - ### Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's. The following manager actions have been added VoicemailBoxSummary - Generate message list for a given mailbox VoicemailRemove - Remove a message from a mailbox folder VoicemailMove - Move a message from one folder to another within a mailbox VoicemailForward - Copy a message from one folder in one mailbox to another folder in another or the same mailbox. - ### app_voicemail: add CLI commands for message manipulation The following CLI commands have been added to app_voicemail voicemail show mailbox Show contents of mailbox @ voicemail remove Remove message from in mailbox @ voicemail move Move message in mailbox & from to voicemail forward Forward message in mailbox @ to mailbox @ - ### sig_analog: Allow immediate fake ring to be suppressed. The immediatering option can now be set to no to suppress the fake audible ringback provided when immediate=yes on FXS channels. Upgrade Notes: Closed Issues: - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms - #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource - #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes - #170: [improvement]: app_voicemail - add CLI commands to manipulate messages - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime - #181: [improvement]: app_voicemail - add manager actions to display and manipulate messages - #202: [improvement]: app_queue: Add support for immediately applying queue caller priority change - #205: [new-feature]: sig_analog: Allow flash to time out to silent hold - #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup - #226: [improvement]: Apply contact_user to incoming calls - #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading - #233: [bug]: Deadlock with MixMonitorMute AMI action - #240: [new-feature]: sig_analog: Add Called Subscriber Held capability - #253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs - #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered" - #263: [bug]: download_externals doesn't always handle versions correctly - #265: [bug]: app_macro isn't locking around channel datastore access - #267: [bug]: ari: refer with display_name key in request body leads to crash - #274: [bug]: Syntax Error in SQL Code - #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement' - #277: [bug]: pbx.c: Compiler error with gcc 12.2 - #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits -- _ -- 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 release 18.20.0
iority change immediately. The 'queue priority caller' CLI command and 'QueueChangePriorityCaller' AMI action now have an 'immediate' argument which allows the caller priority change to be reflected immediately, causing the position of a caller to move within the queue depending on the priorities of the other callers. - ### Adds manager actions to allow move/remove/forward individual messages in a particular mailbox folder. The forward command can be used to copy a message within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry, required to retrieve message ID's. The following manager actions have been added VoicemailBoxSummary - Generate message list for a given mailbox VoicemailRemove - Remove a message from a mailbox folder VoicemailMove - Move a message from one folder to another within a mailbox VoicemailForward - Copy a message from one folder in one mailbox to another folder in another or the same mailbox. - ### app_voicemail: add CLI commands for message manipulation The following CLI commands have been added to app_voicemail voicemail show mailbox Show contents of mailbox @ voicemail remove Remove message from in mailbox @ voicemail move Move message in mailbox & from to voicemail forward Forward message in mailbox @ to mailbox @ - ### sig_analog: Allow immediate fake ring to be suppressed. The immediatering option can now be set to no to suppress the fake audible ringback provided when immediate=yes on FXS channels. Upgrade Notes: Closed Issues: - #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on aarch64 Debian platforms - #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to some resource - #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when immediate=yes - #170: [improvement]: app_voicemail - add CLI commands to manipulate messages - #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime - #181: [improvement]: app_voicemail - add manager actions to display and manipulate messages - #202: [improvement]: app_queue: Add support for immediately applying queue caller priority change - #205: [new-feature]: sig_analog: Allow flash to time out to silent hold - #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup - #226: [improvement]: Apply contact_user to incoming calls - #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading - #233: [bug]: Deadlock with MixMonitorMute AMI action - #240: [new-feature]: sig_analog: Add Called Subscriber Held capability - #253: app_gosub patch appear to have broken predial handlers that utilize macros that call gosubs - #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules registered" - #263: [bug]: download_externals doesn't always handle versions correctly - #265: [bug]: app_macro isn't locking around channel datastore access - #267: [bug]: ari: refer with display_name key in request body leads to crash - #274: [bug]: Syntax Error in SQL Code - #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99 -Wdeclaration-after-statement' - #277: [bug]: pbx.c: Compiler error with gcc 12.2 - #281: [bug]: app_dial: Infinite loop if called channel hangs up while receiving digits -- _ -- 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] Deleting voicemail by program
John, that is some serious script-fu! I does exactly what I was going to do
in perl. However, my initial testing indicates that asterisk will renumber
voicemail boxes to eliminate holes. But I'm still testing.
Thanks again,
Mike.
On Tuesday, October 10, 2023 11:47:35 AM EDT John Harragin wrote:
> Here is something I wrote years ago. I expect you can adjust it for your
> needs
>
>
>
> # cat remove_blank_vmail
> #!/bin/bash
> # remove_blank_vmail takes arguments as voicemail boxes and removes
> messages with audio files shorter then MINSIZE (in bytes)
> #--
> # Description:
> # Author: John Harragin Monroe-Woodbury CSD
> # Created at: Thu Nov 6 12:27:35 EST 2008
> #
> # Copyright: None. Modify and use however you like...
> #
> #--
> # Configure section:
>
> BASEDIR=/var/spool/asterisk/voicemail/default/ # default
> context
> MINSIZE=12000 # 1.5
> seconds
>
> #--subroutines:
>
> ProcessDir () {
> lastfile=""
> delcnt=0
> for file in $(ls -A ${msgdir}/msg*.txt 2>/dev/null); do # the
> redirect supresses msg when dir empty
>if [ $(stat --format=%s ${file/.txt/.wav}) -lt ${MINSIZE} ]; then
> rm ${file/.txt/.*}
> let delcnt++
>fi
>lastfile=${file}
> done
> if [ $delcnt -gt 0 ]; then echo "$delcnt short messages deleted from
> ${msgdir}"; fi
> partfilename=${lastfile/*\/msg/} # get number
> from file name
> highest=${partfilename/.txt/}
> while [[ $highest = 0* ]]; do highest=${highest#0}; done # bash does
> not like leading zeros
> if [ ${#highest} -eq 0 ]; then highest=0; fi # ...or
> blanks for math
> realcount=0
> for ((x=0;x<=${highest};x+=1)); do
>chkname=msg$(printf "%04d" $x) # build name
> - pad with zeros...
>if [ -e ${msgdir}/${chkname}.txt ]; then
> if [ $realcount -ne $x ];then
>newname=msg$(printf "%04d" $realcount)
>for idivfile in $(ls -A ${msgdir}/${chkname}.*); do
> mv ${idivfile} ${msgdir}/${newname}.${idivfile/*\/*./}
>done
> fi
> let realcount++
>fi
> done
> }
>
> #--main:
>
> for ext in "$@"; do
> if [ -d ${BASEDIR}${ext} ];then
>for msgdir in $(ls -d ${BASEDIR}${ext}/*); do
> ProcessDir ${msgdir}
>done
> else
>echo "${BASEDIR}${ext} is not a valid directory"
> fi
> echo "Processed extension $ext"
> done
>
> On Mon, Oct 9, 2023 at 3:06 PM Mike Diehl wrote:
> > Hi all,
> >
> > I need to be able to delete a voicemail message using a program.
> >
> > Is is sufficient to simply delete the .wav and .txt files in the spool
> > directory?
> > Or do I need to also renumber the remaining files?
> >
> > For example, let say a given mailbox has 20 messages in it and I want to
> > delete message number 5. Can I just delete the 2 files and expect that
> > asterisk will renumber them? Or do I need to?
> >
> > Also, is the answer the same when I migrate to storing voicemails in a
> > database?
> >
> > Thanks in advance.
> >
> > Mike
> >
> >
> >
> > --
> > _____
> > -- 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
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Re: [asterisk-users] Deleting voicemail by program
Here is something I wrote years ago. I expect you can adjust it for your
needs
# cat remove_blank_vmail
#!/bin/bash
# remove_blank_vmail takes arguments as voicemail boxes and removes
messages with audio files shorter then MINSIZE (in bytes)
#--
# Description:
# Author: John Harragin Monroe-Woodbury CSD
# Created at: Thu Nov 6 12:27:35 EST 2008
#
# Copyright: None. Modify and use however you like...
#
#--
# Configure section:
BASEDIR=/var/spool/asterisk/voicemail/default/ # default
context
MINSIZE=12000 # 1.5
seconds
#--subroutines:
ProcessDir () {
lastfile=""
delcnt=0
for file in $(ls -A ${msgdir}/msg*.txt 2>/dev/null); do # the
redirect supresses msg when dir empty
if [ $(stat --format=%s ${file/.txt/.wav}) -lt ${MINSIZE} ]; then
rm ${file/.txt/.*}
let delcnt++
fi
lastfile=${file}
done
if [ $delcnt -gt 0 ]; then echo "$delcnt short messages deleted from
${msgdir}"; fi
partfilename=${lastfile/*\/msg/} # get number
from file name
highest=${partfilename/.txt/}
while [[ $highest = 0* ]]; do highest=${highest#0}; done # bash does
not like leading zeros
if [ ${#highest} -eq 0 ]; then highest=0; fi # ...or
blanks for math
realcount=0
for ((x=0;x<=${highest};x+=1)); do
chkname=msg$(printf "%04d" $x) # build name
- pad with zeros...
if [ -e ${msgdir}/${chkname}.txt ]; then
if [ $realcount -ne $x ];then
newname=msg$(printf "%04d" $realcount)
for idivfile in $(ls -A ${msgdir}/${chkname}.*); do
mv ${idivfile} ${msgdir}/${newname}.${idivfile/*\/*./}
done
fi
let realcount++
fi
done
}
#--main:
for ext in "$@"; do
if [ -d ${BASEDIR}${ext} ];then
for msgdir in $(ls -d ${BASEDIR}${ext}/*); do
ProcessDir ${msgdir}
done
else
echo "${BASEDIR}${ext} is not a valid directory"
fi
echo "Processed extension $ext"
done
On Mon, Oct 9, 2023 at 3:06 PM Mike Diehl wrote:
> Hi all,
>
> I need to be able to delete a voicemail message using a program.
>
> Is is sufficient to simply delete the .wav and .txt files in the spool
> directory?
> Or do I need to also renumber the remaining files?
>
> For example, let say a given mailbox has 20 messages in it and I want to
> delete message number 5. Can I just delete the 2 files and expect that
> asterisk will renumber them? Or do I need to?
>
> Also, is the answer the same when I migrate to storing voicemails in a
> database?
>
> Thanks in advance.
>
> Mike
>
>
>
> --
> _
> -- 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
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Re: [asterisk-users] Deleting voicemail by program
On Monday 09 October 2023 at 21:05:55, Mike Diehl wrote: > Hi all, > > I need to be able to delete a voicemail message using a program. > > Is is sufficient to simply delete the .wav and .txt files in the spool > directory? Or do I need to also renumber the remaining files? My approach in a situation like this would often be "try it and see". Leave yourself three voicemail messages, remove the middle one simply by deleting the files, and see what Asterisk makes of what's left behind: - does it report three messages but only play two? - does it report either two or three messages but can only play the first? - does it report two messages and play them without problem? - does it report two messages and fail to play anything? - does it report no messages? - does it have a problem when a fourth message gets left? None of the above behaviour is _necessarily_ transferrable to a future version of Asterisk, but at least it tells you what your current version does when you interfere with in this way.. Antony. -- .evah I serutangis sseltniop tsom eht fo eno eb tsum sihT 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
Re: [asterisk-users] Deleting voicemail by program
The script included with Asterisk (messages-expire.pl) deletes older messages and then renumbers the rest of the messages. I guess you need to do the same. On 09/10/23 2:24 PM, Mike Diehl wrote: Unfortunately, I'm using a version of asterisk that is old enough to not benefit from this... Mike. On Monday, October 9, 2023 3:15:45 PM EDT Michael Bradeen wrote: Hi Mike, New AMI actions were recently added to app_voicemail to let you remotely manipulate a mailbox: https://github.com/asterisk/asterisk/issues/181 Hope this helps. BR, -Mike On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl wrote: Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I need to? Also, is the answer the same when I migrate to storing voicemails in a database? Thanks in advance. Mike -- _ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 -- _ -- 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] Deleting voicemail by program
Unfortunately, I'm using a version of asterisk that is old enough to not benefit from this... Mike. On Monday, October 9, 2023 3:15:45 PM EDT Michael Bradeen wrote: > Hi Mike, > > New AMI actions were recently added to app_voicemail to let you remotely > manipulate a mailbox: > https://github.com/asterisk/asterisk/issues/181 > > Hope this helps. > > BR, > -Mike > > On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl wrote: > > Hi all, > > > > I need to be able to delete a voicemail message using a program. > > > > Is is sufficient to simply delete the .wav and .txt files in the spool > > directory? > > Or do I need to also renumber the remaining files? > > > > For example, let say a given mailbox has 20 messages in it and I want to > > delete message number 5. Can I just delete the 2 files and expect that > > asterisk will renumber them? Or do I need to? > > > > Also, is the answer the same when I migrate to storing voicemails in a > > database? > > > > Thanks in advance. > > > > Mike > > > > > > > > -- > > _ > > -- 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 -- _ -- 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] Deleting voicemail by program
Hi Mike, New AMI actions were recently added to app_voicemail to let you remotely manipulate a mailbox: https://github.com/asterisk/asterisk/issues/181 Hope this helps. BR, -Mike On Mon, Oct 9, 2023 at 1:06 PM Mike Diehl wrote: > Hi all, > > I need to be able to delete a voicemail message using a program. > > Is is sufficient to simply delete the .wav and .txt files in the spool > directory? > Or do I need to also renumber the remaining files? > > For example, let say a given mailbox has 20 messages in it and I want to > delete message number 5. Can I just delete the 2 files and expect that > asterisk will renumber them? Or do I need to? > > Also, is the answer the same when I migrate to storing voicemails in a > database? > > Thanks in advance. > > Mike > > > > -- > _ > -- 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 -- _ -- 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] Deleting voicemail by program
Hi all, I need to be able to delete a voicemail message using a program. Is is sufficient to simply delete the .wav and .txt files in the spool directory? Or do I need to also renumber the remaining files? For example, let say a given mailbox has 20 messages in it and I want to delete message number 5. Can I just delete the 2 files and expect that asterisk will renumber them? Or do I need to? Also, is the answer the same when I migrate to storing voicemails in a database? Thanks in advance. Mike -- _ -- 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] CDR gets lost
Yes - update your my.conf to increase the timeouts by a large amount, then restart mysql daemon. Here's some details: https://telium.io/en/topic/mysql-server-has-gone-away/ From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Federico Sent: Tuesday, September 19, 2023 11:36 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] CDR gets lost I noticed that of Asterisk is idle many hours, then the CDR (with batch=yes) does not get written to the MySQL database anymore. Is there a keepalive command for ODBC? cdr show status Call Detail Record (CDR) settings -- Logging:Enabled Mode: Simple Log calls by default: Yes Log unanswered calls: Yes Log congestion: Yes Ignore bridging changes:No Ignore dial state changes: No * Registered Backends --- Adaptive ODBC cdr_manager (suspended) cdr-custom csv -- _ -- 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] CDR gets lost
I noticed that of Asterisk is idle many hours, then the CDR (with batch=yes) does not get written to the MySQL database anymore. Is there a keepalive command for ODBC? cdr show status Call Detail Record (CDR) settings -- Logging:Enabled Mode: Simple Log calls by default: Yes Log unanswered calls: Yes Log congestion: Yes Ignore bridging changes:No Ignore dial state changes: No * Registered Backends --- Adaptive ODBC cdr_manager (suspended) cdr-custom csv -- _ -- 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 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Hello Jerry, when you run asterisk using su, ownership of audio device files does not get updated. When you login, you get the permissions. You can verify by ls -l and getfacl on the device file. Marek --- Original Message --- On Thursday, September 14th, 2023 at 14:33, Jerry Geis wrote: > On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis wrote: > >>>An issue[1] was already created by asterisk at phreaknet.org and they also >>>put >>>a fix up for review and inclusion[2]. >> >>>[1] https://github.com/asterisk/asterisk/issues/308 >>>[2] https://github.com/asterisk/asterisk/pull/309 >> >> The change "seems" to be working. >> Will test more tomorrow - had to leave. >> THANKS! >> Jerry > > Yes - this fix is working for me. > > Only issue I have now is, I used to run asterisk like this: > su silentm -c "/usr/sbin/asterisk -fn" > I also tried > su silentm -l -c "/usr/sbin/asterisk -fn" > > these do not work for the chan_console. I have to actually login as silentm > and then run asterisks - to HEAR the audio. > doing su above I do not hear the audio - but the CLI looks the same - no > errors. > > Thoughts? > > 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On Wed, Sep 13, 2023 at 5:20 PM Jerry Geis wrote: > >An issue[1] was already created by asterisk at phreaknet.org and they > also put > >a fix up for review and inclusion[2]. > > >[1] https://github.com/asterisk/asterisk/issues/308 > >[2] https://github.com/asterisk/asterisk/pull/309 > > > The change "seems" to be working. > Will test more tomorrow - had to leave. > THANKS! > > Jerry > Yes - this fix is working for me. Only issue I have now is, I used to run asterisk like this: su silentm -c "/usr/sbin/asterisk -fn" I also tried su silentm -l -c "/usr/sbin/asterisk -fn" these do not work for the chan_console. I have to actually login as silentm and then run asterisks - to HEAR the audio. doing su above I do not hear the audio - but the CLI looks the same - no errors. Thoughts? 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>An issue[1] was already created by asterisk at phreaknet.org and they also put >a fix up for review and inclusion[2]. >[1] https://github.com/asterisk/asterisk/issues/308 >[2] https://github.com/asterisk/asterisk/pull/309 The change "seems" to be working. Will test more tomorrow - had to leave. 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
An issue[1] was already created by aster...@phreaknet.org and they also put
a fix up for review and inclusion[2].
[1] https://github.com/asterisk/asterisk/issues/308
[2] https://github.com/asterisk/asterisk/pull/309
On Wed, Sep 13, 2023 at 4:27 PM Jerry Geis wrote:
>
> I have found that I can add the restart of asterisk (killall -9 asterisk)
> to the h extension- BOY is that UGLY.
>
> chan_console is not a testing device - how can we get this nasty bug fixed
> ?
>
> Jerry
> --
> _
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>
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> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
I have found that I can add the restart of asterisk (killall -9 asterisk) to the h extension- BOY is that UGLY. chan_console is not a testing device - how can we get this nasty bug fixed ? 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
> After a hung call, can you run core restart now from the asterisk console? Doing a "killall -9 asterisk" is the only thing that works I tried killall asterisk - does not free up the channel the asterisk "core restart now" takes like a good 20 seconds to return but does work. The issue is I cannot run it after teh Dial() as the Dial(Console/default,20,g) never returns to the dial plan. 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
It worked with my test. I'm on Asterisk 18.19.0
-- Executing [517xxx@voipms:4] System("IAX2/voipms-15815", "asterisk
-rx 'core restart now'") in new stack
-- Remote UNIX connection
Asterisk uncleanly ending (0).
Executing last minute cleanups
== Destroying musiconhold processes
== Manager unregistered action DBGet
== Manager unregistered action DBGetTree
== Manager unregistered action DBPut
== Manager unregistered action DBDel
== Manager unregistered action DBDelTree
Preparing for Asterisk restart...
Asterisk is now restarting...
asterisk*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
After a hung call, can you run core restart now from the asterisk console?
Doug
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>Using system() you could issue a asterisk -rx 'core restart now' So I tried this exten => s,1,Playback(beep) exten => s,n,Dial(Console/default,20,g) exten => s,n,Hangup exten => s,n,System(asterisk -rx 'core restart now') But it does not continue. Last thing I see is "Exited non zero" so its not doing the hangup or the system. 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
I have noticed that once my message speaks - the server thinks its done and HUNGUP, the endpoint STILL thinks the channel is active - the last message says "Rx: BYE" on sip show channels I tried ADDING to Dial() ,20,g and then had a Hangup after teh dial. Its NOT getting there to hangup. 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>> Is there a dial plan call that can "exit asterisk" or completely reload >> everything - killall active calls and start over ? Using system() you could issue a asterisk -rx 'core restart now' Doug -- _ -- 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 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Is there a dial plan call that can "exit asterisk" or completely reload everything - killall active calls and start over ? seems the console/dummy (chan_console) is holding some resource. How do I just "exit" and start over after call came in ? 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
I don't know if this will help you, but looking back through an old config I have for an older version of Asterisk, I had used chan_console with the old and now defunct app_rpt app to listen to audio on various nodes via the console for testing. Here is what I did: In console.conf, I defined this: [default] input_device = default output_device = default autoanswer = no context = extension = callerid = language = en overridecontext = no mohintrepret = default active = yes In modules.conf I loaded the audio module (in this case it was chan_alsa.so, but I also could use chan_oss.so). I made sure noload was commented out for chan_alsa.so In alsa.conf, I defined some of the same things as in console.conf: [general] autoanswer=no context= extension= inputdevice=plughw:0,1 otuputdevice=plughw:0,0 mute=true You'll need to check your ALSA device to see what the input and output devices are. That last line is important, since on the console you may not have a mic that works to talk, you just want to listen, In extensions.conf, I defined a dialplan that instead of trying to dial out, it just answered the call and then threw me into the app. Then to dial from the console, I woudl use: console dial And it woudl use the context I defined and launch the Rpt app. What you could do is define somehting like this ,but have the extension use DISA so that you can then get dumped into your normal dialplan logic where you could use "console dial xxx". No guarantees that this will work with a newer version of Asterisk, but this did work with a 1.8 setup I used to have (that I have the configs saved for). -Stacy On 9/8/23 10:28 AM, Jerry Geis wrote: So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
So I have done through chan_console.c and searched for console_pct_lock() - every one - has a matching console_pvt_unlock() How is the deadlock occurring ? 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] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>>> How do we get this working For the time being, go back to 18.14.0 Doug -- _ -- 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 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>
>
> Not sure if this is the same thing you're seeing, but chan_console
> currently has a known deadlock issue that has not been resolved:
> https://issues-archive.asterisk.org/ASTERISK-30481
> It's quite easy to reproduce, so it may be the case that the module is
> currently unusable.
>
Well this is a bummer
[Sep 8 08:45:28] WARNING[313684][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
--- <("<) --- Hangup on Console --- (>")> ---
[Sep 8 08:45:28] ERROR[313683]: utils.c:1027 lock_info_destroy: Thread
'stream_monitor started at [ 390] chan_console.c start_stream()'
still has a lock! - 'pvt' (0x55e647a245e0) from 'stream_monitor' in
chan_console.c:281!
How do we get this working
Jerry
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
On 9/8/2023 8:18 AM, Jerry Geis wrote:
But when I do a second test. Asterisk HANGS on ChanIsAvail()
Then I thought lets SKIP that - and just let it do the Dial() - I
stopped everything - got it running again. - and then the Dial() hangs
on the second call.
So both ChanIsAvail() or Dial() both hang on the second call in.
So only 1 call in will work.
Below is the CLI report of the call that works.
This is my context
[smvoice-mediacontroller-public-address]
exten => s,1,ChanIsAvail(Console/default)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/default)
exten => s,n,Hangup
Not sure if this is the same thing you're seeing, but chan_console
currently has a known deadlock issue that has not been resolved:
https://issues-archive.asterisk.org/ASTERISK-30481
It's quite easy to reproduce, so it may be the case that the module is
currently unusable.
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
Some progress to report:
I had to run asterisk as the user logged in - actually not even that. I
could not "su user -c " to that user - I had to run it as that user.
Then I did a test and got audio! Great...
But when I do a second test. Asterisk HANGS on ChanIsAvail()
Then I thought lets SKIP that - and just let it do the Dial() - I stopped
everything - got it running again. - and then the Dial() hangs on the
second call.
So both ChanIsAvail() or Dial() both hang on the second call in.
So only 1 call in will work.
Below is the CLI report of the call that works.
This is my context
[smvoice-mediacontroller-public-address]
exten => s,1,ChanIsAvail(Console/default)
exten => s,n,GotoIf($["${AVAILCHAN}" = ""]?smvoice-busy,s,1)
exten => s,n,Playback(beep)
exten => s,n,Dial(Console/default)
exten => s,n,Hangup
Now what ???
Jerry
onnected to Asterisk 18.18.0 currently running on nuc11cdev2 (pid = 282195)
== Using SIP RTP CoS mark 5
> 0x7feeec0086b0 -- Strict RTP learning after remote address set to:
192.168.1.8:17526
-- Executing [public_address@smvoice-mediacontroller:1]
SoftHangup("SIP/devgeis_to_nuc11cdev2-", "ALSA/dummy") in new stack
-- Executing [public_address@smvoice-mediacontroller:2]
Goto("SIP/devgeis_to_nuc11cdev2-",
"smvoice-mediacontroller-public-address,s,1") in new stack
-- Goto (smvoice-mediacontroller-public-address,s,1)
-- Executing [s@smvoice-mediacontroller-public-address:1]
NoOp("SIP/devgeis_to_nuc11cdev2-", "JERRY") in new stack
-- Executing [s@smvoice-mediacontroller-public-address:2]
Playback("SIP/devgeis_to_nuc11cdev2-", "beep") in new stack
> 0x7feeec0086b0 -- Strict RTP switching to RTP target address
192.168.1.8:17526 as source
-- Playing 'beep.gsm' (language
'en')
-- Executing [s@smvoice-mediacontroller-public-address:3]
Dial("SIP/devgeis_to_nuc11cdev2-", "Console/default") in new stack
--- <("<) --- Call to device 'default' on console from 'MyName Here'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
-- Called Console/default
-- Console/default answered SIP/devgeis_to_nuc11cdev2-
-- Channel Console/default joined 'simple_bridge' basic-bridge
[Sep 8 08:07:10] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
-- Channel SIP/devgeis_to_nuc11cdev2- joined 'simple_bridge'
basic-bridge
> 0x7feeec0086b0 -- Strict RTP learning complete - Locking on source
address 192.168.1.8:17526
-- Channel SIP/devgeis_to_nuc11cdev2- left 'simple_bridge'
basic-bridge
-- Channel Console/default left 'simple_bridge' basic-bridge
[Sep 8 08:07:17] WARNING[282457][C-0001]: chan_console.c:651
console_indicate: Don't know how to display condition 26 on Console/default
--- <("<) --- Hangup on Console --- (>")> ---
== Spawn extension (smvoice-mediacontroller-public-address, s, 3) exited
non-zero on 'SIP/devgeis_to_nuc11cdev2-'
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
In the old days when I was using console/dsp, I would have to use
alsamix from the console to verify that the output wasn't muted. Maybe
it's still the same.
Doug
On 9/7/23 03:43 PM, Jerry Geis wrote:
ok switching to "Console/default" does show the text
--- <("<) --- Call to device 'default' on console from 'default'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
However I don't hear any audio.
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