Have you tried using the EVAL function?
On Tue, Jan 24, 2023, 7:38 PM
wrote:
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>
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On Wed, Dec 7, 2022 at 11:34 AM Joshua C. Colp wrote:
> On Wed, Dec 7, 2022 at 11:26 AM Dan Cropp wrote:
>
>> On two VMs, we encounter a strange behavior when we upgrade from 18.12.1
>> to 18.15.0 (also tried 18.15.1 last night).
>>
>> When we roll the VMs back to 18.12.1, we don’t see the behav
On Wed, Dec 7, 2022 at 11:26 AM Dan Cropp wrote:
> On two VMs, we encounter a strange behavior when we upgrade from 18.12.1
> to 18.15.0 (also tried 18.15.1 last night).
>
> When we roll the VMs back to 18.12.1, we don’t see the behavior repeat.
>
>
>
> We have a Kamailio VM front ending the aste
On two VMs, we encounter a strange behavior when we upgrade from 18.12.1 to
18.15.0 (also tried 18.15.1 last night).
When we roll the VMs back to 18.12.1, we don't see the behavior repeat.
We have a Kamailio VM front ending the asterisk.
It sends OPTIONS messages periodically.
After startup (an
The Asterisk Development Team would like to announce the release of
Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves
issues r
On Wed, Nov 30, 2022 at 6:03 PM TTT wrote:
> I’ve noticed on several occasions that if Asterisk starts without a
> network connection, then even if the network connection is restored, DNS
> lookups fail.
>
>
>
> After the connection is restored I can successfully do NSLOOKUPs from the
> command l
I've noticed on several occasions that if Asterisk starts without a network
connection, then even if the network connection is restored, DNS lookups
fail.
After the connection is restored I can successfully do NSLOOKUPs from the
command line, but the IAX2 registration attempts keep failing beca
Hi am using asterisk 18.14.0 with pulse audio and dialing console dsp and
getting a warble or a clipping in my audio.
This is my cli log
== Using SIP RTP CoS mark 5
> 0x7f47b80132a0 -- Strict RTP learning after remote address set to:
192.168.1.8:19436
-- Executing [public_address@smvo
I am running ubuntu 20.04 fully patched along with Asterisk 18.8.0
This is a VM environment with VMWare.
I found this in the logs today.
[1768362.083207] CPU: 2 PID: 1939739 Comm: asterisk Tainted: G
OEL5.15.0-52-generic #58~20.04.1-Ubuntu
[1768362.083209] call_cpuidle+0x23/0x50
[1768362.08
The Asterisk Development Team would like to announce the release of Asterisk
20.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 20.0.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
19.7.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.7.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.15.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.15.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
16.29.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.29.0 resolves several issues reported by the
community and would have not been
ANyone ever ran into a situation when Call coming from Call Manager into
asterisk, is successful coming across - but no Audio ???
But then the next call - audio is heard - its once in a great while no
audio - most time it works.
Anything I might look for ? How do I debug that?
Thanks
jerry
--
I am just doing a basic call in.
exten => 140,1,Answer
exten => 140,n,Playback(beep)
exten => 140,n,Dial(MulticastRTP/basic/239.168.4.90:3040//t(15))
exten => 140,n,Hangup
this works - but "sometimes" I get reports that "nothing" was heard.
Is there anything special to do for multicast ?
Any thou
Hi Josh
> This was a security issue[1] which was solved.
>
> [1] https://downloads.asterisk.org/pub/security/AST-2021-006.html
Thanks, filing Bugreport with Debian, hopefully they will push 16.16.2
to security updates.
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G-
On Tue, Aug 23, 2022 at 12:30 PM Benoit Panizzon
wrote:
> Hi List
>
> I can reproduce Asterisk 16.16.1 segfaulting in this situation:
>
> Asterisk configured with Application "ReceiveFax".
>
> Incoming call with SDP:
>
> v=0
> o=prt-cbl-sbc1 1418830458 1418830459 IN IP4 157.161.X.X
> s=sip call
>
Hi List
I can reproduce Asterisk 16.16.1 segfaulting in this situation:
Asterisk configured with Application "ReceiveFax".
Incoming call with SDP:
v=0
o=prt-cbl-sbc1 1418830458 1418830459 IN IP4 157.161.X.X
s=sip call
c=IN IP4 157.161.X.X
t=0 0
m=audio 11828 RTP/AVP 9 8 101
a=rtpmap:9 g722/8000
The Asterisk Development Team would like to announce the release of Asterisk
19.6.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.6.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.14.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.14.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
16.28.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.28.0 resolves several issues reported by the
community and would have not been
Hi Joel,
We see some channels stuck on the StopMonitor application which could be
the cause, though there are only about 20 of them.
An Asterisk restart is the immediate cure for sure - we were hoping to
identify some cause in order to prevent a repeat.
Thanks very much for your advice.
On Thu
I would check if you don't have any channels in a hung/zombie state...
Have a look if "core show calls" matches "core show channels".
Either way, it seems wonky, so you might end up having to give that
asterisk a restart... :S
On Wed, Jul 27, 2022 at 6:21 PM David Cunningham
wrote:
> Hello,
>
Hello,
We have an Asterisk 13.38.2 server which today started giving "we couldn't
allocate a port for RTP" errors. The output of "netstat -anp" showed that
Asterisk was using all 10,000 ports allocated for RTP, even though it only
had a maximum of around 200 concurrent calls at any point in the da
On 2022-07-18 20:05, Joshua C. Colp wrote:
On Mon, Jul 18, 2022 at 9:02 AM John Covici
wrote:
I am using freepbx latest 16 version -- am I subject to this
problem?
I am not using elastics, but I installed on a Debian bullseye
server,
so this is of definite concern to me.
Thanks.
The forum p
On 2022-07-18 20:02, John Covici wrote:
I am using freepbx latest 16 version -- am I subject to this problem?
I am not using elastics, but I installed on a Debian bullseye server,
so this is of definite concern to me.
Thanks.
On Mon, 18 Jul 2022 06:45:41 -0400,
Joshua C. Colp wrote:
[1 ]
[1.
On 2022-07-18 18:45, Joshua C. Colp wrote:
On Mon, Jul 18, 2022 at 7:43 AM Turritopsis Dohrnii Teo En Ming
wrote:
Dear Joshua Colp,
Noted with thanks. So the vulnerability is not related to the
Asterisk
open source project at all?
It is not. The vulnerability mentioned is regarding FreePBX
On Mon, Jul 18, 2022 at 9:02 AM John Covici wrote:
> I am using freepbx latest 16 version -- am I subject to this problem?
> I am not using elastics, but I installed on a Debian bullseye server,
> so this is of definite concern to me.
>
> Thanks.
>
The forum post I linked is where this is being
I am using freepbx latest 16 version -- am I subject to this problem?
I am not using elastics, but I installed on a Debian bullseye server,
so this is of definite concern to me.
Thanks.
On Mon, 18 Jul 2022 06:45:41 -0400,
Joshua C. Colp wrote:
>
> [1 ]
> [1.1 ]
> On Mon, Jul 18, 2022 at 7:43 A
On Mon, Jul 18, 2022 at 7:43 AM Turritopsis Dohrnii Teo En Ming <
c...@teo-en-ming.com> wrote:
>
> Dear Joshua Colp,
>
> Noted with thanks. So the vulnerability is not related to the Asterisk
> open source project at all?
>
It is not. The vulnerability mentioned is regarding FreePBX and Elastix,
On 2022-07-18 17:01, Joshua C. Colp wrote:
On Mon, Jul 18, 2022 at 4:11 AM Turritopsis Dohrnii Teo En Ming
wrote:
Subject: Asterisk IP PBX VoIP Servers Hacked by Hackers
Good day from Singapore,
I am sharing the following news articles for more awareness.
News article #1: Hackers Targeting
On Mon, Jul 18, 2022 at 4:11 AM Turritopsis Dohrnii Teo En Ming <
c...@teo-en-ming.com> wrote:
> Subject: Asterisk IP PBX VoIP Servers Hacked by Hackers
>
> Good day from Singapore,
>
> I am sharing the following news articles for more awareness.
>
> News article #1: Hackers Targeting VoIP Servers
Subject: Asterisk IP PBX VoIP Servers Hacked by Hackers
Good day from Singapore,
I am sharing the following news articles for more awareness.
News article #1: Hackers Targeting VoIP Servers By Exploiting Digium
Phone Software
Link:
https://thehackernews.com/2022/07/hackers-targeting-voip-serv
The Asterisk Development Team would like to announce the release of Asterisk
19.5.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.5.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.13.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.13.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
16.27.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.27.0 resolves several issues reported by the
community and would have not been
Hello,
I'm trying to figure out how blind transfers are supposed to work with ARI.
When two channels are bridged together through ARI, and one of them
performs a blind SIP transfer, two things happen :
- a Local channel is instanciated and goes through the dialplan at the
specified destinati
On Fri, May 20, 2022 at 1:43 PM Dan Cropp wrote:
> We have a customer where their switch sends pidf+xml presence information
> in the SIP INVITE message.
>
>
>
> Does Asterisk process this pidf+xml information?
>
> Does it store this in a channel variable that a dial plan could access?
>
> If not
We have a customer where their switch sends pidf+xml presence information in
the SIP INVITE message.
Does Asterisk process this pidf+xml information?
Does it store this in a channel variable that a dial plan could access?
If not, does it store present this information to AMI/ARI applications in a
Also figured that out.
For chan sip, specifying the ptime in the allow statement
allow=g722:20,alaw:20 made the ptime header be present and the SBC
happy.
--
Mit freundlichen Grüssen
-Benoît Panizzon- @ HomeOffice und normal erreichbar
--
I m p r o W a r e A G-Leiter Commerce Kunden
Hi List
Just ran into another weird issue...
In Swiss Telephone Interconnection, ptime=20 is a requirement.
So on our SBC we enforce the presence of ptime=20 to avoid issues.
I have an asterisk with chan_sip in the LAB which behaves weirdly...
Inbound SDP audio part:
m=audio 15542 RTP/AVP 9 8
The Asterisk Development Team would like to announce the release of Asterisk
19.4.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.4.1 resolves an issue reported by the
community and would have not been possibl
The Asterisk Development Team would like to announce the release of Asterisk
18.12.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.12.1 resolves an issue reported by the
community and would have not been possi
The Asterisk Development Team would like to announce the release of Asterisk
16.26.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.26.1 resolves an issue reported by the
community and would have not been possi
On Fri, May 13, 2022 at 1:05 PM Dan Cropp wrote:
> I have been using Asterisk 18.11.2.
>
> Just tried Asterisk 18.12.0 and am running into a problem with the
> res_pjsip_transport_websocket.
>
>
>
> Using Ubuntu 20
>
> I use a bash shell script to compile Asterisk with settings.
>
> I didn’t modi
I have been using Asterisk 18.11.2.
Just tried Asterisk 18.12.0 and am running into a problem with the
res_pjsip_transport_websocket.
Using Ubuntu 20
I use a bash shell script to compile Asterisk with settings.
I didn't modify any settings from Asterisk 18.11.2 build that works.
After compiling,
The Asterisk Development Team would like to announce the release of Asterisk
19.4.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.4.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.12.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.12.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
16.26.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.26.0 resolves several issues reported by the
community and would have not been
I need help that how to compile asterisk version 16.x with backtrace flag
, debug or coredump flags. I tried many times but it has not been compiled
successfully.
Would be thankful if someone could help us how to use these flags in
compilation that work with asterisk 16.x successfully.
i am usin
The Asterisk Development Team would like to announce the release of Asterisk
19.3.3.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.3.3 resolves an issue reported by the
community and would have not been possibl
The Asterisk Development Team would like to announce the release of Asterisk
18.11.3.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.11.3 resolves an issue reported by the
community and would have not been possi
The Asterisk Development Team would like to announce the release of Asterisk
16.25.3.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.25.3 resolves an issue reported by the
community and would have not been possi
The Asterisk Development Team would like to announce security releases for
Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases are
released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
These releases are available for immediate download at
https://downloads.asteri
The Asterisk Development Team would like to announce the release of Asterisk
19.3.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.3.1 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.11.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.11.1 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
16.25.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.25.1 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
19.3.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.3.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.11.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.11.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
16.25.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.25.0 resolves several issues reported by the
community and would have not been
Le 08/03/2022 à 03:40, TTT a écrit :
I have a fresh Asterisk 18 install on a fresh OS (AWS Linux 2). I
used the service file from contrib directory and commented out user
and group settings so it runs under root.
[...]
Hello,
The service file provided in the "contrib" directory defines a
I have a fresh Asterisk 18 install on a fresh OS (AWS Linux 2). I used the
service file from contrib directory and commented out user and group
settings so it runs under root.
When I start Asterisk via systemd it waits a long time and then times out.
Reporting failure to the command line. Res
The Asterisk Development Team would like to announce security releases for
Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases are
released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
These releases are available for immediate download at
https://downloads.asteri
The Asterisk Development Team would like to announce the release of Asterisk
19.2.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.2.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.10.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.10.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
16.24.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.24.0 resolves several issues reported by the
community and would have not been
We're having a problem where Asterisk 16 refuses to play voicemail
recordings and greetings stored in wav49 format. It throws an error
similar to the following:
2022-01-27 11:31:37 format_wav.c: Not a supported wav file format
(49). Only PCM encoded, 16 bit, mono, 8kHz/16kHz files are sup
OK, that tells me something, I will disable pjsit for now, learn about
it and try again.
On Sun, 09 Jan 2022 06:39:55 -0500,
John Harragin wrote:
>
> [1 ]
> [1.1 ]
> You can also set up multiple physical or vlan(ed) interfaces and bind sip
> to one and pjsip to the other - then you have to set
> On 9/01/2022, at 7:11 PM, John Covici wrote:
>
> On Sat, 08 Jan 2022 19:17:57 -0500,
> Antony Stone wrote:
>>
>>> On Sunday 09 January 2022 at 00:50:27, John Covici wrote:
>>>
>>> Hi. I am using asterisk 18.3 and freepbx.
>>
>> Hm, which version of FreePBX uses Asterisk 18.3?
>>
>>> H
On Sat, 08 Jan 2022 19:17:57 -0500,
Antony Stone wrote:
>
> On Sunday 09 January 2022 at 00:50:27, John Covici wrote:
>
> > Hi. I am using asterisk 18.3 and freepbx.
>
> Hm, which version of FreePBX uses Asterisk 18.3?
>
> > How can both sip and pjsip be listening at port 5060 at the same time
On Sunday 09 January 2022 at 00:50:27, John Covici wrote:
> Hi. I am using asterisk 18.3 and freepbx.
Hm, which version of FreePBX uses Asterisk 18.3?
> How can both sip and pjsip be listening at port 5060 at the same time
They can't.
One might be on TCP and the other on UDP, but you can't ha
On Sat, 8 Jan 2022, John Covici wrote:
How can both sip and pjsip be listening at port 5060 at the same time...
They can't. One application per address/port pair.
You can configure pjsip to bind to another address and/or port while you
figure it out the configuration.
--
Thanks in advance,
Hi. I am using asterisk 18.3 and freepbx. How can both sip and pjsip
be listening at port 5060 at the same time, for instance I get:
[2022-01-08 17:08:59] SECURITY[244351] res_security_log.c:
SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity="Error",Service="PJSIP",EventV
This includes Jira, the Wiki and Gerrit due to loss of internet access. If
you're currently signed in to the community forums, you're OK but new
logins won't be accepted. The IRC channels are unaffected.
Probably has something to do with this in Huntsville:
[image: image.png]
--
George Joseph
>>> asterisk doesn't support .ogg file format (digged through
Yes it does, if it's complled in with it.
Under make menuselect
=> Format Interpreters
You'll see the development libraries that need to be installed before
re-compiling for ogg playback support
Doug
--
asterisk doesn't support .ogg file format (digged through
apps/app_playback.c, main/file.c)
dev*CLI> core show file formats
Format Name Extensions
-- --
slin mp3 mp3
slin48 ogg_opus opus
so with
/var/lib/asterisk/sounds/output-ogg.opus i
>>> exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output-ogg)
If the actual filename is output.ogg then the code should be
exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output)
You'll also need to confirm that you compiled Asterisk with Vorbis support.
Doug
--
modified
exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output.ogg)
changed to
exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output-ogg)
file
[root@dev cz]# file /var/lib/asterisk/sounds/output-ogg.ogg
/var/lib/asterisk/sounds/output-ogg.ogg: Ogg data
[Dec 22 14:39:26] WARNI
>>> exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output.ogg,)
Do not use the .ogg when describing the filename.
Doug
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Aster
hi,
is it possible to playback ogg/opus files to alaw sip clients?
exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output.ogg,)
[Dec 22 14:05:53] WARNING[49275][C-0004]: file.c:789
ast_openstream_full: File /var/lib/asterisk/sounds/output.ogg does not
exist in any format
[Dec 22
Hi Daniel,
This is a production server which is running well over years (asterisk
11-13-16) and this happend with the latest version. Only valid option
you gave is the core show locks. I ask the list before opening a bug
report, as usually.
Please don't let the fact that the system has bee
Hi,
1) You should change your name on your email client so it doesn't say
"Administrator"
2) Please follow the instructions at
https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source
3) Compile with DEBUG_THREADS and DONT_OPTIMIZE, but note this will
incur a performance
The Asterisk Development Team would like to announce the release of Asterisk
19.1.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.1.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.9.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.9.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
16.23.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.23.0 resolves several issues reported by the
community and would have not been
On Sat, Nov 13, 2021 at 9:41 AM Jerry Geis wrote:
> I am trying to use the SIPML5 at
> https://www.doubango.org/sipml5/call.htm?svn=252
> and when I hit the login button - and asterisk says "wrong password" and
> the web page says Forbidden.
>
> I have triple checked that I entered the correct pa
I am trying to use the SIPML5 at
https://www.doubango.org/sipml5/call.htm?svn=252
and when I hit the login button - and asterisk says "wrong password" and
the web page says Forbidden.
I have triple checked that I entered the correct password on the website, I
can see the password on Asterisk sip.c
hi,
i'm testing
https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Configuration_res_prometheus
anybody who can share grafana dashboard json ;) ?
thanks
Marek
--
_
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Le 08/11/2021 à 18:10, Jerry Geis a écrit :
[...]
Hi Jean
interesting - was not aware of the unicastrtp channel - been looking for more
information on it - not finding much.
Is there anyway to bring "in" audio with unicastrtp. I can perhaps see
'sending" audio out - but I'm looking for both di
>Hello,
>You may use a UnicastRTP channel. It allows you to specify an IP/port to
>connect to.
>Regards, Jean Aunis
Hi Jean
interesting - was not aware of the unicastrtp channel - been looking
for more information on it - not finding much.
Is there anyway to bring "in" audio with unicastrtp.
Hi -
I have a device that has 16 RTP ports. I desire to bring that audio
into Asterisk... is that possible ?
The device does not run SIP at all just RTP audio. I am using Asterisk 18.
How might I do that ?
Thanks,
Jerry
Hello,
You may use a UnicastRTP channel. It allows you to specify an
-boun...@lists.digium.com] On Behalf
Of Jerry Geis
Sent: Sunday, November 7, 2021 8:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk bring in RTP audio
Hi -
I have a device that has 16 RTP ports. I desire to bring that audio into
Asterisk
Hi -
I have a device that has 16 RTP ports. I desire to bring that audio into
Asterisk... is that possible ?
The device does not run SIP at all just RTP audio. I am using Asterisk 18.
How might I do that ?
Thanks,
Jerry
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The Asterisk Development Team would like to announce the release of Asterisk
19.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 19.0.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
18.8.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 18.8.0 resolves several issues reported by the
community and would have not been p
The Asterisk Development Team would like to announce the release of Asterisk
16.22.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.22.0 resolves several issues reported by the
community and would have not been
This turned out to be a brain fart on my part, not a bug in Asterisk.
Thanks for your help and sorry to waste your time ...
Cheers,
Kingsley.
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On Fri, 2021-10-22 at 11:11 -0300, Joshua C. Colp wrote:
> I don't provide direct support like that. As there seems to be a bug
> and you have a case that reproduces it with logs, then you can file
> an issue[1] and the current individual doing bug triage will look. If
> it is accepted there is no
On Fri, Oct 22, 2021 at 11:07 AM Kingsley Tart wrote:
> Hi,
>
> I have built a new Asterisk installation:
>
> root@gw9:/tmp# asterisk -V
> Asterisk 18.7.1
>
> It still does the same thing, which is
>
> a. Asterisk receives INVITE containing SDP telephone-event
> b. Asterisk uses Dial with pjsip a
Hi,
I have built a new Asterisk installation:
root@gw9:/tmp# asterisk -V
Asterisk 18.7.1
It still does the same thing, which is
a. Asterisk receives INVITE containing SDP telephone-event
b. Asterisk uses Dial with pjsip and sends INVITE to destination
including SDP telehone-event
c. Asterisk re
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