The Asterisk Development Team would like to announce
the release of asterisk-21.1.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.1.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
This release resolves issues
The Asterisk Development Team would like to announce
the release of asterisk-20.6.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.6.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
This release resolves issues
The Asterisk Development Team would like to announce
the release of asterisk-18.21.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.21.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
This release resolves issues
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
--
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
>
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
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
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
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)
>
>
>
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
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
> 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,
>>>
>>>
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
--
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
>> >
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
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
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
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
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
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
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
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:
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
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
Hello Federico,
Can you please review the Bug Report requirements, and submit a new bug
report for this issue?
https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/
Also Note:
Before filing a bug report... Your issue may not be a bug or could have
been fixed already. Run
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
>
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
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
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)?
-
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
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
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
2023-11-08 18:14:13] ERROR[571246][C-17e2] : Got 19 backtrace records
# 0: [0x5bd18a] asterisk utils.c:2800 __ast_assert_failed()
# 1: [0x4618e3] asterisk astobj2.c:589 __ao2_ref()
# 2: [0x58e660] asterisk stasis_cache.c:824 update_create()
# 3: [0x58efed] asterisk stasis_cache.c:903
Hello List
I would very much like to have some feedback on this. Where do I have to
look ? Is it in the Asterisk version (13.38.3) maybe ? Is it for sure in
my config ?!
Kind regards.
Op 28/06/2023 om 16:14 schreef Jonas Kellens:
Hello list
when trying to set up webRTC
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
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
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,
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
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
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
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
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
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,
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
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:
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
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
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
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
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
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
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
>
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
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.
--
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
--
The Asterisk Development Team would like to announce
the release of asterisk-21.0.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/21.0.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
This release resolves issues
The Asterisk Development Team would like to announce
the release of asterisk-20.5.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/20.5.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
This release resolves issues
The Asterisk Development Team would like to announce
the release of asterisk-18.20.0.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/18.20.0
and
https://downloads.asterisk.org/pub/telephony/asterisk
This release resolves issues
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
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)
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
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...
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:
>
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
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
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,
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:
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
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
>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
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
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
> 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
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
>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
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
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>> 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
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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
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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
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
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>>> How do we get this working
For the time being, go back to 18.14.0
Doug
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Check out the new Asterisk community forum at:
>
>
> 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
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()
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
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
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