examples of "interesting" information like ICE result and howto make
"minimal" configuration of pjproject.conf
i.e.
for debugging app_queue.so
core set debug 5 app_queue.so
for debugging RTP
core set debug 10 rtp_engine
core set debug 10 res_rtp_asterisk
rtp set debug on
logger.conf
rtp
On Thu, Dec 12, 2019 at 10:57 AM marek wrote:
> thank you very much. this is exactly whats needed for debug
>
> example output for your info
> [Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
> icess0x7f5d44081e88 .Added new remote candidate from the request:
> 2.2.2.2:57536
> [Dec 12
thank you very much. this is exactly whats needed for debug
example output for your info
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
icess0x7f5d44081e88 .Added new remote candidate from the request:
2.2.2.2:57536
[Dec 12 15:39:19] DEBUG[2182][C-]: pjproject: :
On Thu, Dec 12, 2019 at 8:57 AM marek wrote:
> Asterisk is on public IP (as described in the first email)
>
> i have 10 years experience in voip, 4 years webrtc in production. i know
> about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism
>
> but i confess. i dont understand
Asterisk is on public IP (as described in the first email)
i have 10 years experience in voip, 4 years webrtc in production. i know
about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism
but i confess. i dont understand WHY Asterisk SOMETIMES switches
destination IP in RTP.
On Thu, Dec 12, 2019 at 7:57 AM marek wrote:
> with wireshark i need decrypt traffic every call which is time consuming.
> get debug from pjnat through asterisk is not possible because of technical
> reasons or nobody did it?
>
>
> in my case its strange that ice candidates are the same
>
> good
with wireshark i need decrypt traffic every call which is time
consuming. get debug from pjnat through asterisk is not possible because
of technical reasons or nobody did it?
in my case its strange that ice candidates are the same
good call
v=0
o=- 3669976329745317845 2 IN IP4 127.0.0.1
s=-
On Thu, Dec 12, 2019 at 6:39 AM marek wrote:
> hi,
>
> i have following topology
>
> PSTN - Asterisk internet - router - jssip client (wss)
>
> Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP
> connection to PSTN
>
> router - public IP/private IP (NAT)
>
> jssip
hi,
i have following topology
PSTN - Asterisk internet - router - jssip client (wss)
Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP
connection to PSTN
router - public IP/private IP (NAT)
jssip client - private IP - sip over websocket to Asterisk PJSIP
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
These releases are available for immediate download at
On Wed, Nov 20, 2019 at 5:40 AM O. Hartmann wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA256
>
> Am Sat, 16 Nov 2019 07:39:08 -0400
> "Joshua C. Colp" schrieb:
>
> > On Sat, Nov 16, 2019 at 4:07 AM O. Hartmann
> wrote:
> >
> > > -BEGIN PGP SIGNED MESSAGE-
> > > Hash: SHA256
>
On Sat, Nov 16, 2019 at 4:07 AM O. Hartmann wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA256
>
>
> Hello,
>
> we're running a small Asterisk appliance on a PCengine APU2C4. Base
> operating system is
> FreeBSD 12-STABLE, most recent incarnation as of today.
>
> Since update of port
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256
Hello,
we're running a small Asterisk appliance on a PCengine APU2C4. Base operating
system is
FreeBSD 12-STABLE, most recent incarnation as of today.
Since update of port net/asterisk16 to the latest bug fix revision 16.6.1, we
face a severe
The Asterisk Development Team would like to announce the release of Asterisk
17.0.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 17.0.0 resolves several issues reported by the
community and would have not been
When we created the External Media addition to ARI we created an
ExternalMedia object to be returned from the channels/externalMedia REST
endpoint. This object contained the channel object that was created plus
local_address and local_port attributes (which are also in the Channel
variables). At
They only problem I have found so far is while trying to install
Alembic for SQLAlchemy (for realtime configs). Those are the only
packages that I cannot get working properly. Vanilla Asterisk works
fine with the only extra package needed being libedit-devel that is not
included in any
At the current time, we do not recommend attempting to build Asterisk on
CentOS 8. Many packages Asterisk uses are not yet available and would
require building from their sources. The Asterisk packages are also not
available in the EPEL 8 or CentOS 8 repositories yet for the same reason.
We'll
On Wed, Oct 16, 2019, at 3:03 PM, Marcelo Garay wrote:
> Thank you for your answer!!
>
> Unfortunately I'm using the CEF browser based on Chromium and it
> doesn't support H264 because license isn't free so renegotiation is not
> an option.
>
> I've noticed when recording a channel with video
The Asterisk Development Team would like to announce the release of Asterisk
16.6.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.6.1 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
13.29.1.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.29.1 resolves several issues reported by the
community and would have not
On Tue, Oct 15, 2019, at 5:14 PM, Marcelo Garay wrote:
> Hello,
>
> I’m trying to record video on a channel (from webrtc) using ARI (POST
> /channels/{channelId}/record), but when I specify h264 for the format I
> get error: ast_writestream: Unable to translate to format h264, source
> format
The Asterisk Development Team would like to announce the release of Asterisk
16.6.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.6.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
13.29.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.29.0 resolves several issues reported by the
community and would have not
On Fri, Oct 4, 2019, at 1:45 AM, Andreas Wehrmann wrote:
>
> On 03/10/2019 16:24, Joshua C. Colp wrote:
> > In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately
> > codec negotiation is not written or implemented in the way you need. There
> > are some hints provided
On 03/10/2019 16:24, Joshua C. Colp wrote:
In PJSIP there is the PJSIP_MEDIA_OFFER dialplan function[1] but ultimately
codec negotiation is not written or implemented in the way you need. There are
some hints provided internally for outgoing legs but the result is still
ultimately
On Thu, Oct 3, 2019, at 11:10 AM, Andreas Wehrmann wrote:
>
> On 03.10.19 15:08, Administrator TOOTAI wrote:
>
> > Before calling the gatreway add
> >
> > same = n,set(SIP_CODEC=alaw)
> >
> > [...]
> >
>
> Hey there,
>
> that doesn't work as it seems to be implemented for chan_sip only;
> I'm
On 03.10.19 15:08, Administrator TOOTAI wrote:
Before calling the gatreway add
same = n,set(SIP_CODEC=alaw)
[...]
Hey there,
that doesn't work as it seems to be implemented for chan_sip only;
I'm using chan_pjsip; sorry if I didn't explain myself properly.
Anyway, in my case that would
Hi
Le 03/10/2019 à 13:13, Andreas Wehrmann a écrit :
[...]
- Even if direct_media is disabled: Is there a way to make Asterisk
always use a common codec between SIP endpoints,
so it doesn't need to transcode?
Before calling the gatreway add
same = n,set(SIP_CODEC=alaw)
[...]
--
Hello people,
I've ran into two problem that I can't seem to be able to solve on my own.
Here's my scenario (running Asterisk 13.28.1):
In short: - Asterisk behaves unexpectedly (at least to me) when
negotiating between endpoints
that have a different but intersecting set of codecs
g doesn't work in
Ast 13.22.0.
Thanks for the help.
Kind regards,
-Original Message-
From: Patrick Laimbock
Sent: Friday, 13 September 2019 13:37
To: viljo...@verishare.co.za; asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Asterisk 13 on Microsoft Azure Centos 7 instance
cannot
Hi Stefan,
> Hi all
>
> I maintain the above - it was set up by an external party with whom relations
> have now been severed by my employer.
>
> Quite early after the deployment it became evident that all .gsm audio files
> produced on this virtual instance at Azure via MixMonitor are
Hi all
I maintain the above - it was set up by an external party with whom relations
have now been severed by my employer.
Quite early after the deployment it became evident that all .gsm audio files
produced on this virtual instance at Azure via MixMonitor are corrupt. If you
play back the
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1,
15.7.4 and 16.5.1.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
In article <874506323.2924334.1567645810...@mail.yahoo.com>,
bilal ghayyad wrote:
>
> Thank you a lot for your kindly help and reply. Actually it helped me a
> lot.I was using _X. in the extensions.conf at
> the trunkinbound context.Can you advise me what is the difference between _X.
> and
Thank you a lot for your kindly help and reply. Actually it helped me a lot.I
was using _X. in the extensions.conf at the trunkinbound context.Can you advise
me what is the difference between _X. and s? In other words, when it is better
to use s and when it is better to use _X.?
Again, I am
On Thu, Aug 22, 2019, at 5:33 AM, Jonas Kellens wrote:
> Hello
>
> I see on the CLI :
>
> tst*CLI> core show hints
> -= Registered Asterisk Dial Plan Hints =-
> 50@blf : SIP/testacc7 State:Idle Watchers 3
> 6001@blf : Custom:q-6001 State:Idle Watchers 1
> 5@blf : SIP/testacc6
Hello
I see on the CLI :
tst*CLI> core show hints
-= Registered Asterisk Dial Plan Hints =-
50@blf : SIP/testacc7
State:Idle Watchers 3
6001@blf : Custom:q-6001 State:Idle
Watchers 1
5@blf
On Sat, Aug 17, 2019, at 3:07 AM, Michael Maier wrote:
> Hello!
>
>
> Few words about the usage of asterisk:
> - 2 registered endpoints
> - 4 SIPS / SRTP trunks
> - 46 calls at 2019-08-15
> - the sip:isp.de trunk hadn't been used
>
>
> Some findings:
>
> - The problem seems to be triggered
Hello!
I encountered an outage of asterisk which showed like that:
- 2019-08-10 07:22:21 Asterisk start
- 2019-08-15 19:39:33 WARNING taskprocessor.c: The
'pjsip/outreg/ispPJSIP-0060' task processor queue reached 500 scheduled
tasks.
- 2019-08-15 19:39:34 WARNING
On Mon, Aug 5, 2019, at 8:35 AM, Fernando Pardo wrote:
> Hello, everybody. I'm migrating a module I've developed for Asterisk 11
> to use it on Asterisk 16. One of the trickiest parts I haven't found a
> way around is the removal of the ast_call_feature struct, which I used
> to execute a
Thank you, dear Asterisk Development Team, for this great software!
> The Asterisk Development Team would like to announce the release of
> Asterisk 13.28.0.
--
_
-- Bandwidth and Colocation Provided by
The Asterisk Development Team would like to announce the release of Asterisk
16.5.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.5.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
13.28.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.28.0 resolves several issues reported by the
community and would have not
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
These releases are available for immediate download at
>> I had that issue at a previous employer and got around it by using ALSA
instead.
Thanks but the other piece I need is forcing me to use pulseaudio. That's
what it connects to.
Jerry
>
--
_
-- Bandwidth and Colocation
>>> I setup and extension to connect me with Console/Dsp. I am hearing the
>>> audio but its warbly or does not sound right. Any thoughts on what I need
>>> to do for that ?
I had that issue at a previous employer and got around it by using ALSA instead.
Doug
--
Hi All,
I am running pulseaudio on my asterisk server.
I setup and extension to connect me with Console/Dsp. I am hearing the
audio but its warbly or does not sound right. Any thoughts on what I need
to do for that ?
Another thought is I tried to setup the Musiconhold to be custom and the
Hi All,
Is there a way to get Airplay music into asterisk ?
I have used Shairport-sync to get Airplay to play audio on my pulseaudio
computer - but was wanting that to come into Asterisk ?
Thanks
Jerry
--
_
-- Bandwidth and
My self-compiled Asterisk also shows that speex dependencies are not installed
Speex Coder/Decoder
Depends on: speex(E), speex_preprocess(E)
Can use: speexdsp(E)
You'll need to installed the dependencies and re-compile.
Doug
--
On Friday 05 July 2019 at 16:33:56, Jerry Geis wrote:
> I have no speex translation
> core show translation paths speex
> --- Translation paths SRC Codec "speex" sample rate 8000 ---
> speex:8000 To slin:8000 : No Translation Path
> Does not look good. no paths... Did something
I have no speex translation
ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24
slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw
ulaw - 9150 15000 1500015000 15000 9000 17000 17000 17000
17000 17000 17000 17000 17000 15000 15000 17250
On Friday 05 July 2019 at 16:03:42, Jerry Geis wrote:
> I think this is what your looking for:
> [Jul 5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find a
> codec translation path: (speex) -> (speex32)
Indeed, it was.
> My linphone side only has speex@32K enabled.
>
> My
I think this is what your looking for:
Found RTP audio format 119
Found audio description format speex for ID 119
Capabilities: us - (speex|speex16|speex32|g722|ulaw|alaw|gsm), peer -
audio=(speex32)/video=(nothing)/text=(nothing), combined - (speex32)
Non-codec capabilities (dtmf): us - 0x1
On Friday 05 July 2019 at 14:22:22, Jerry Geis wrote:
> Hi all - I am using asterisk 13.27.0 with Linphone.
> I turned off all codes on linphone except the one I want to try. For
> example:
> opus and speex (so only one enabled at a time).
> Then did this same on asterisk for the linphone
Hi all - I am using asterisk 13.27.0 with Linphone.
I turned off all codes on linphone except the one I want to try. For
example:
opus and speex (so only one enabled at a time).
Then did this same on asterisk for the linphone extension.
disallow=all
allow=speex
(for example).
Then I place my
On Tue, Jul 2, 2019, at 6:34 AM, Mahipal Singh wrote:
> Hello all,
> I am using node ari client library and, when i dial call using mobile
> and my mobile is showing that it is in dialling state, but i receive
> 'StasisStart' event.
> Actually my code is according that whenever i received
that.
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Israel Gottlieb
Sent: Monday, July 1, 2019 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 179, Issue 1
how about sticking in a pbx
how about sticking in a pbx between [c] and [h]
so when [h] hangsup you send to [s] if that is 3rd party else i dont see
how you could redirect [c] at all
else maybe ask them to have [h] redirect [c] to [s] then [h] will also be
out of the call
On Mon, Jul 1, 2019, 20:03 Send asterisk-users
The Asterisk Development Team would like to announce the release of Asterisk
16.4.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.4.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
13.27.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.27.0 resolves several issues reported by the
community and would have not
On Thu, May 30, 2019, at 11:30 AM, Jonas Kellens wrote:
> Hello
>
> is this mailing list still active ?
Seems like it. :D I responded previously. Many people have moved to
Discourse[1] though and it sees more activity.
[1] https://community.asterisk.org/
--
Joshua C. Colp
Digium - A Sangoma
Hello
is this mailing list still active ?
Op 10-05-19 om 14:10 schreef Jonas Kellens:
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving
On Sat, May 25, 2019, at 2:34 PM, Saint Michael wrote:
> Joshua
> Is there a way in PJSIP to send the audio between the parties always,
> unless one of the parties is behind a NAT?
> A session refresh would work.
> That my only problem with PJSIP. This is routine in the old sip channel.
Any such
On Tue, May 28, 2019, at 9:56 AM, Jonas Kellens wrote:
> Hello
>
> is this mailing list still active ?
It is still active. Video under chan_sip, however, is not something many do and
in particular it is possible with WebRTC that something has changed and caused
problems or there is a bug in a
Hello
is this mailing list still active ?
Op 10-05-19 om 14:10 schreef Jonas Kellens:
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving
Joshua
Is there a way in PJSIP to send the audio between the parties always,
unless one of the parties is behind a NAT?
A session refresh would work.
That my only problem with PJSIP. This is routine in the old sip channel.
On Sat, May 25, 2019 at 1:03 PM
wrote:
> Send asterisk-users mailing
Hello
I am trying to set up webRTC video calls from my Chrome webbrowser
(Fedora) to my Chrome webbrowser (Windows 10).
There is local video input (I can see myself), but never video on the
receiving side.
This is the case in both directions (so it makes no difference which
peer is
The Asterisk Development Team would like to announce the release of Asterisk
16.3.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 16.3.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team would like to announce the release of Asterisk
13.26.0.
This release is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.26.0 resolves several issues reported by the
community and would have not
On Tue, Apr 2, 2019, at 9:06 PM, Sungtae Kim wrote:
>
> On 4/3/19 1:29 AM, Joshua C. Colp wrote:
> > On Tue, Apr 2, 2019, at 8:26 PM, Matthew Jordan wrote:
> >>
> >> On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp wrote:
> >>> On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> >>> > I get the
On 4/3/19 1:29 AM, Joshua C. Colp wrote:
On Tue, Apr 2, 2019, at 8:26 PM, Matthew Jordan wrote:
On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp wrote:
On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> I get the desired use case to run app_amd from within a Stasis
> application, but I’m
On Tue, Apr 2, 2019, at 8:26 PM, Matthew Jordan wrote:
>
>
> On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp wrote:
> > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> > > I get the desired use case to run app_amd from within a Stasis
> > > application, but I’m not sure about app_queue.
On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp wrote:
> On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> > I get the desired use case to run app_amd from within a Stasis
> > application, but I’m not sure about app_queue. You have everything at
> > your disposal within ARI itself to replicate
On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote:
> I get the desired use case to run app_amd from within a Stasis
> application, but I’m not sure about app_queue. You have everything at
> your disposal within ARI itself to replicate all of the functionality
> of app_queue and beyond.
Yes,
Um... Because... I can?
Tbh, no reason... app_queue is just my favorite module. :P
On 4/2/19 11:41 PM, Joshua C. Colp wrote:
On Tue, Apr 2, 2019, at 6:20 PM, Sungtae Kim wrote:
Hi Asterisk users,
I'm one of Asterisk ARI users, and trying to designing the new ARI for
application execution in
On 4/2/19 11:41 PM, Joshua C. Colp wrote:
On Tue, Apr 2, 2019, at 6:20 PM, Sungtae Kim wrote:
Hi Asterisk users,
I'm one of Asterisk ARI users, and trying to designing the new ARI for
application execution in Stasis().
This will be made possible for executing the applications in the
Stasis()
On Tue, Apr 2, 2019, at 6:20 PM, Sungtae Kim wrote:
> Hi Asterisk users,
>
> I'm one of Asterisk ARI users, and trying to designing the new ARI for
> application execution in Stasis().
>
> This will be made possible for executing the applications in the
> Stasis() application.
>
> But, before
Hi Asterisk users,
I'm one of Asterisk ARI users, and trying to designing the new ARI for
application execution in Stasis().
This will be made possible for executing the applications in the
Stasis() application.
But, before going further, I would like to know which application needs
to be
On Thu, Mar 28, 2019, at 4:56 PM, Dan Cropp wrote:
> Hi Joshua,
>
> Unfortunately, I tried including the Refer-Sub true and also false in
> the REFER packet and Cisco seems to ignore them.
>
> Refer-Sub: false
> and
> Refer-Sub: true
>
> The only thing that seems to work properly with the
)?
-Original Message-
From: Dan Cropp
Sent: Thursday, March 28, 2019 12:55 PM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Asterisk Transfers
Thank you Joshua.
We're trying to run more tests.
We believe Cisco may not be adhering to the specification. Unfortunately
On Thu, Mar 28, 2019, at 2:56 PM, Dan Cropp wrote:
> Thank you Joshua.
>
> We're trying to run more tests.
> We believe Cisco may not be adhering to the specification.
> Unfortunately, we're also stuck with having to make it work.
>
> An interesting test, I commented out the norefersub from
the lack of a Refer-Sub header in the REFER
incorrectly?
Dan
-Original Message-
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Thursday, March 28, 2019 9:18 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Transfers
On Thu, Mar 28, 2019, at 11:10 AM, Dan
On Thu, Mar 28, 2019, at 11:10 AM, Dan Cropp wrote:
>
> Is there no one who knows if there is a way to turn off the norefersub
> setting?
>
>
> Supported: norefersub
>
>
> This happens in the TRYing, OK, and other commands in response to the INVITE.
>
>
> For chan_sip, I noticed it does
, Ringing, OK inside them. This basically
gives the chan_sip code the ability to know if the REFER (Transfer) is
succeeding or not.
Dan
From: asterisk-users On Behalf Of Dan
Cropp
Sent: Monday, March 25, 2019 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk
Does anyone know if there is a way to disable the norefersub for PJSIP?
It appears this is causing problems with a test we're running with Cisco.
A wireshark trace from a system where the transfer with Cisco works versus a
trace with Asterisk/Cisco shows one big difference being the supported:
Hi,
I found this:
https://lists.kamailio.org/pipermail/sr-users/2019-January/104312.html
It turns out my issue was caused by a wrong *nat=* setting for the device...
After changing:
nat=force_rport,comedia
to:
nat=comedia
I can now see correct IP and RTT in asterisk `sip show peers` for
Hello,
We have a couple asterisk11 servers behind a Kamailio4 proxy. We are in the
process of upgrading to asterisk16 and Kamailio5 and I'm testing out Path:
support with chan_sip (migration to PJSIP is not possible right now due to
integrations with other systems).
Functionality-wise things are
On Tue, Mar 12, 2019, at 3:05 AM, Stefan Viljoen wrote:
> Hi Joshua
>
> Does the survey imply that there are big changes coming for Asterisk?
>
> E. g. features or facilities will be dropped / deprecated from the open
> source version in new releases, big changes to existing facilities /
>
Hi Joshua
Does the survey imply that there are big changes coming for Asterisk?
E. g. features or facilities will be dropped / deprecated from the open source
version in new releases, big changes to existing facilities / protocols, what
is supported officialy by Digium for the official
On Mon, Mar 11, 2019, at 7:16 AM, Administrator TOOTAI wrote:
> Le 11/03/2019 à 10:23, Marcelo Terres a écrit :
> > Hello Jean-Denis.
> >
> > I believe the idea is that you answer the survey for each type of
> > scenarios you are running.
> >
> > So one for call centre, another one for ivr,
Le 11/03/2019 à 10:23, Marcelo Terres a écrit :
Hello Jean-Denis.
I believe the idea is that you answer the survey for each type of
scenarios you are running.
So one for call centre, another one for ivr, etc...
And what for instance about exact version of asterisk?
We are in the same
Hello Jean-Denis.
I believe the idea is that you answer the survey for each type of scenarios
you are running.
So one for call centre, another one for ivr, etc...
Regards,
Marcelo
On Mon, 11 Mar 2019, 02:10 Jean-Denis Girard, wrote:
> Hi Matt,
>
> I would have loved to participate to the
Hi Matt,
I would have loved to participate to the survey, but I feel it does
apply to my situation: as an integrator, I'm installing Asterisk for
call centers, PBX, IVR... so I can not answer the first question of the
survey ;) I also have dfferent versions installed.
This is not a negative
Hey All,
For those of you that do not know me, my name is Matthew Fredrickson
and I’m the project lead for the Asterisk project. First off, I wanted
to thank all of you that contribute in various ways to the project –
whether it be at a developmental level, answering questions on forums
and
> 1. What is the content of ${OPERATOR}?
>
> 2. What do you have for this connection in sip.conf?
>
> 3. What number/s have you been assigned by your upstream SIP provider?
>
> Antony.
Hello
The problem appeared in siptrunk. The problem is "insecure=very". This
"insecure=invite" improved.
Very
On Tuesday 05 March 2019 at 17:22:16, Gokan Atmaca wrote:
> > exten => _13XXX,1,dial(${OPERATOR},20)
1. What is the content of ${OPERATOR}?
2. What do you have for this connection in sip.conf?
3. What number/s have you been assigned by your upstream SIP provider?
Antony.
> On Tue, Mar 5,
> exten => _13XXX,1,dial(${OPERATOR},20)
Hello
"SIP/2.0 401 Unauthorized" Unfortunately the negative. An asterisk
indicates a 404 error.
On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle wrote:
>
> On 3/5/19 2:46 AM, Gokan Atmaca wrote:
> > Asterisk can send calls, but I don't get a call. What
On 3/5/19 2:46 AM, Gokan Atmaca wrote:
Asterisk can send calls, but I don't get a call. What could be the problem?
[from-siptrunk]
exten => 13XXX,1,dial(${OPERATOR},20)
You are trying to match a pattern, so this needs to be
exten => _13XXX,1,dial(${OPERATOR},20)
Doug
--
Hello
Asterisk can send calls, but I don't get a call. What could be the problem?
[from-siptrunk]
exten => 13XXX,1,dial(${OPERATOR},20)
Thanks.
--
_
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The Asterisk Development Team would like to announce security releases for
Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
16.2.1.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases
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