on this older software to get into a bad state to cause the issues with AMI and
Asterisk state?
Dan
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a description of the 12-byte packet header
fields?
Dan
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that works is externalmedia)
Dan
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core set debug category off rtcp
rtcp set debug off
rtp set debug off
Dan
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x7f777b4ba609 in start_thread (arg=) at
pthread_create.c:477
#21 0x7f777b23c133 in clone () at
../sysdeps/unix/sysv/linux/x86_64/clone.S:95
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Wednesday, August 9, 2023 1:31 PM
To: Asterisk Users Mailing List - Non-Commercial Dis
, Aug 9, 2023 at 3:20 PM Dan Cropp
mailto:dcr...@amtelco.com>> wrote:
I have a customer who just encountered a crash while running Asterisk 18.17.1
version.
I’m trying to adapt to the changes so not sure where best to look or how to
possibly report this.
I started by going through
es_pjsip_nat.c:470
#7 nat_on_tx_message (tdata=0x7f773c633ac8) at res_pjsip_nat.c:479
#8 0x7f777bc6fc66 in ?? ()
#9 0x7f777bc71610 in ?? ()
#10 0x7f773c633ac8 in ?? ()
#11 0x55e7c1ac7708 in ?? ()
#12 0x7f777bc77c69
Discussion
Subject: Re: [External] [asterisk-users] [External] Asterisk rtp.conf stunaddr
setting - what happens if there is an outage
On Tue, Feb 7, 2023 at 11:18 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Thank you Joshua.
Going back to your idea of the ice_host_candidates. (Again, apo
3.4,include_local_address
Using this, would we no longer need the stunaddr configured?
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Monday, February 6, 2023 4:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] Aster
I7955a046293f913ba121bbd82153b04439e3465f require the dnsmgr.conf to be enabled?
Dan
From: Dan Cropp
Sent: Monday, February 6, 2023 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Asterisk rtp.conf stunaddr setting - what happens if there is an outage
Over the weekend, we had several
covery?
Is there a recommendation on how to prevent this from happening?
Any thoughts?
Dan
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of Channel ARI
requests that are allowed when the call is not handed off to the Stasis
application
On Mon, Jan 30, 2023 at 7:30 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
We have used AMI for many years and I’m in the process of migrating to ARI.
My understanding is the call should be
like I am on the right track for migrating from AMI to Stasis,
ARI/Websocket support?
Dan
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Please disregard, I figured out what I was doing wrong.
Dan
From: Dan Cropp
Sent: Friday, January 20, 2023 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Question on ARI externalMedia
A couple years ago, I know I had ARI externalMedia working. Trying to figure
umber":""},"connected":{"name":"","number":""},"accountcode":"","dialplan":{"context":"default","exten":"s","priority":1,"app_name":"AppDial
We use the sox (SoX - Sound eXchange) package to perform many audio
manipulation routines our customers require.
Dan
-Original Message-
From: asterisk-users On Behalf Of
astuserl...@mytelpbx.com
Sent: Friday, December 9, 2022 5:53 AM
To: asterisk-users@lists.digium.com
Subject
it working, I will create an issue report.
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Wednesday, December 7, 2022 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] Asterisk 18.12.1 to 18.15.0 upgrade
seems to have
.12.10
Was there some configuration change introduced after 18.12.1 that I missed?
Any thoughts?
Dan
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Asking because I see there is a new DeadlockStart event added to 18.15.0 but
the AMI_VERSION value is still 7.0.2
Dan
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) is sent.
On some occasions the agent is incorrect or makes a mistake in doing this too
soon, so the transfer fails.
At that point, add everyone back into the ConfBridge, waiting for the agent to
decide when to retry.
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Thursday, September 1
indicating when the early media ends and the call is really Up?
Dan
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New
Thank you for the explanation George.
This makes it very easy to know if the Geo Location is the configured or an
incoming value.
Dan
From: asterisk-users On Behalf Of
George Joseph
Sent: Thursday, August 25, 2022 7:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
to
prefer_incoming even when it should be discard_config or prefer_config.
same => n,Set(MY__GEO_PROFILE_PRECEDENCE=${GEOLOC_PROFILE(profile_precedence)})
Dan
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party passed a field/value that doesn’t match the Asterisk defaults,
our software will add the GEOLOC_PROFILE(name) to the Originate Variable field.
Then I send the Originate packet to Asterisk via AMI.
Thank you for all your work on this!!!
Dan
From: asterisk-users On Behalf Of
George Joseph
@192.168.12.34
Exten: createcall
Context: mycontext
Priority: 1
Timeout: 6
CallerID: John Smith <8005551212>
Variable: PJSIP_HEADER(add,abc)=123,CALLERID(num-pres)=allowed_passed_screen
Async: true
Codecs: ulaw
Dan
From: Dan Cropp
Sent: Monday, August 15, 2022 2:00 PM
To: Asterisk Users Mailin
ccountCode: 20^M
Context: IS^M
Exten: s^M
Priority: 1^M
Uniqueid: 1660588901.0^M
Linkedid: 1660588901.0^M
Variable: GEOLOCPROFILESTATUS^M
Value: 0^M
^M
Dan
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C
Thank you Joshua
rc2 resolved the issue I was seeing.
However, it sounds like it would be best for me to configure the location_info
without the leading underscore for the variable name.
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Friday, August 12, 2022 8:04 AM
To: Asterisk
those variables?
Dan
From: asterisk-users On Behalf Of
George Joseph
Sent: Wednesday, August 10, 2022 8:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] [External] [External] Geo location
18.14.0-rc1 question
Sorry for the delay but this
Joseph
mailto:gjos...@sangoma.com>> wrote:
On Tue, Aug 2, 2022 at 1:13 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Is the allow_routing setting on the geolocation Wiki Profile also not fully
implemented?
Well, 99% of the code is there. The 1% is parsing the config option. Not
Is the allow_routing setting on the geolocation Wiki Profile also not fully
implemented?
In the code, I see geolocation_routing used instead of allow_routing.
Tried both and Asterisk indicates it cannot find suitable setting so it doesn’t
create the profile object.
Dan
From: Dan Cropp
Sent
Thank you George.
From: asterisk-users On Behalf Of
George Joseph
Sent: Tuesday, August 2, 2022 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] Geo location 18.14.0-rc1 question
On Tue, Aug 2, 2022 at 8:46 AM Dan Cropp
mailto:d
ansmission_allowed=no
location_reference = IS_loc_22
location_refinement = ROOM=292
location_refinement = FLR=1
Dan
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to toggle between 911 call and the
patients call.
It sounds like the inherited channel variables is exactly what I was looking
for. External application just passes the replacement values for these
variables (_HNO in the example you used, along with many others).
Have an awesome day!
Dan
way to achieve this?
Alternatively, I could generate an internal local channel, configure the
GeoLocProfile on it, configure all GEOLOC_PROFILE adjustments on it, then have
it perform the Dial. If the other end answers or not, treat it exactly as we
currently do using the Originate.
Dan
, when
they join the conference bridge it is actually going. Thus, if the bridge
options had the record enabled it would start recording.
If only marked user joins first, it's met the criteria and will conference and
start recording.
Dan
-Original Message-
From: asterisk-users On Behalf
Thank you Joshua.
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Monday, May 23, 2022 3:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] [External] Geolocation/E911
On Mon, May 23, 2022 at 5:52 PM Dan Cropp
mailto:d
availability requirements (911 call center). Asterisk would be far
cleaner since there would be no timing issues of call information from 2 boxes.
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Monday, May 23, 2022 3:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
calls to work.
Dan
From: asterisk-users On Behalf Of
Sebastian Nielsen
Sent: Monday, May 23, 2022 3:19 PM
To: 'Mailing List'
Subject: Re: [External] [asterisk-users] Geolocation/E911
What are you talking about exactly?
What I have understand, E911 geo information is sent “out of band” when
to be worked on)?
Dan
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Thank you Joshua.
Thank you to those working on this addition.
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Friday, May 20, 2022 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] [External] Asterisk PJSIP pidf+xml
] Asterisk PJSIP pidf+xml presence
question
On Fri, May 20, 2022 at 1:43 PM Dan Cropp
mailto:d...@amtelco.com>> 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
in any
way?
At Astricon 2019, one of the presenters talking about Presence and the growing
requirements for 911. Needing to know not just building, but where (floor,
etc) in building.
I don't recall the full presentation, but I'm guessing pidf is what he was
referring to.
Dan
mf_mode = rfc4733
webrtc = yes
disallow = all
allow = ulaw
transport = transport3
acl = acl5
Might this be because PJSIP 2.12 changes to the
“WebRTC updates with AEC3 & AGC2”
From: Dan Cropp
Sent: Friday, May 13, 2022 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
] [External] [External] Asterisk 18.12.0
question
On Fri, May 13, 2022 at 3:19 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Thanks Joshua.
I didn’t describe that very well.
When I first noticed the res_http_transport_websocket wasn’t loading on that
box, I compared the modules folder o
-Commercial Discussion
Subject: Re: [External] [asterisk-users] [External] Asterisk 18.12.0 question
On Fri, May 13, 2022 at 2:43 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Hi Joshua,
Thank you for helping me diagnose this.
Interesting that they are the exact same between versions.
File
disable the chan_sip web support in the configuration
files.
Dan
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New to
on.
Did something change with the res_pjsip_transport_websocket where it requires
something new?
Does PJSIP 2.12 require something new that the previous PJSIP version Asterisk
used did not require?
Dan
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As far as I'm aware Josh, it doesnt stop a call from happening - I've had
the same "errors" pop up when using Twilio and Simwood but calls continue
just fine.
On Thu, Dec 2, 2021 at 2:30 PM Joshua C. Colp wrote:
> On Thu, Dec 2, 2021 at 10:18 AM James Cloos wrote:
>
>> > "KT" == Kingsley
It shouldnt stop the call from happening. It will be something else... up
your debugging level and see what else you get
Lots of providers go against this part of the spec but I've run Asterisk 18
with twilio over sip over tls and everything worked, it just spat out the
error line
On Thu, Dec
)
On Thu, Nov 18, 2021 at 4:34 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
We currently use the Queue. Under app_queue, it uses module res_monitor (which
is on the to be deprecated list).
Is it safe to continue using Queue (app_queue)?
The app_queue module is not on the deprecated list, the
We currently use the Queue. Under app_queue, it uses module res_monitor (which
is on the to be deprecated list).
Is it safe to continue using Queue (app_queue)?
Dan
This email is intended only for the use of the party to which it is addressed
and may contain information that is privileged
Thank you George.
Using slin16 instead of generic slin resolved the issue.
Have a good day!
Dan
From: asterisk-users On Behalf Of
George Joseph
Sent: Wednesday, October 13, 2021 8:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [External] Re: [asterisk-users
interested in a2519b4b-4d90-4d18-906b-717d02f8d569
Have a good day!
Dan
This email is intended only for the use of the party to which it is addressed
and may contain information that is privileged, confidential, or protected by
law. If you are not the intended recipient you are hereby notified
When we perform ExternalMedia with the slin format, we are still receiving ulaw
rtp packets. Asterisk logs show it's selecting ulaw.
I'm guessing we are missing a menuselect or configuration setting.
Anyone have any suggestions for the possible cause and what to look at?
Dan
This email
We are running Asterisk 16.17.0 and discovered what we think is an issue.
We have a single call in a ConfBridge.
Tell the ConfBridge to start recording.
We see non-stop audiohook.c 160 samples failures. As soon as we stop recording
(AMI ConfBridgeStopRecord) these failures stop.
[10/04
-Commercial Discussion
Subject: Re: [asterisk-users] Any thoughts on Asterisk 16.17.0 outputting FRACK
refcount related messages
On Thu, Sep 23, 2021 at 1:59 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
We have an extremely busy/large customer. They run fine most of the time, but
period
.
"WARNING[20710][C-33a6] channel.c: Exceptionally long voice queue length
queuing to CBAnn/"
Is this expected?
Any thoughts on how to resolve this?
Dan
This email is intended only for the use of the party to which it is addressed
and may contain information that is
/libpthread.so.0(+0x76db) [0x7fc1e9ebf6db]
# 7: /lib/x86_64-linux-gnu/libc.so.6(clone+0x3f) [0x7fc1e93f971f]
Dan
This email is intended only for the use of the party to which it is addressed
and may contain information that is privileged, confidential, or protected by
law. If you
10450500
pjsip/distributor-0b07
2902 0 8450500
Any thoughts?
Dan
From: Dan Cropp
Sent: Tuesday, September 14, 2021 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
control.
Dan
From: asterisk-users On Behalf Of
George Joseph
Sent: Tuesday, September 14, 2021 12:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Large system seeing single CPU core spiking
On Tue, Sep 14, 2021 at 9:19 AM Dan Cropp
mailto:d
:07 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
I am working with a very large customer running Asterisk with PJSIP. Systems
total channels have been over 2500 (which includes hundreds of local channels
and ConfBridges) when the issues occur.
It’s running on a Hyper-V VM with 12 CPU
?
Any thoughts on how to diagnose the problem?
Any other thoughts/comments?
Dan
This email is intended only for the use of the party to which it is addressed
and may contain information that is privileged, confidential, or protected by
law. If you are not the intended recipient you are hereby
We found no way to do this from AMI.
Very tied up on other projects, but if another developer wanted to look into
adding support for it, I believe it would be something along these lines….
int action_hold(struct mansession *s, const struct message *m)
{
const char *channelarg =
, but that is not a very nice scheme for live
transcription.
Any other suggestions for a better way to do this?
Dan
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ding to connect another local channel to one of the
ConfBridges and start recording. Unfortunately, it's a scenario we are stuck
with due to a unique customer requirement.
Any thoughts or suggestions?
Dan
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?
Dan
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New to Asterisk? Start here:
https://wiki.asterisk.org/wiki
Please disregard. I found my problem. We use a unique folder for the spool.
Once I created the recording folder in our directory everything worked as
expected.
Dan
From: asterisk-users On Behalf Of Dan
Cropp
Sent: Tuesday, August 11, 2020 9:24 AM
To: 'asterisk-users@lists.digium.com
nitor
drwxrwxrwx 2 root root 4096 Sep 13 2019 recording
drwxr-xr-x 2 root root 4096 Sep 13 2019 system
drwxr-xr-x 2 root root 4096 Sep 13 2019 tmp
drwxr-xr-x 2 root root 4096 Sep 13 2019 voicemail
Any suggestions on what I am doing wrong?
Thank you Joshua. That matches what I experienced last week.
I will build the string to play the number prompts using individual sound
prompts instead of using number.
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Monday, August 10, 2020 8:49 AM
To: Asterisk Users Mailing List
Thank you Jöran
That did the trick.
I had been trying to figure out how to do this without the json content and
couldn’t figure out how to do it.
Dan
From: asterisk-users On Behalf Of
Jöran Vinzens
Sent: Monday, August 10, 2020 8:59 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi Jöran,
Would it be possible to see an example using curl of how you are passing the
PAI Header through ARI create?
Dan
From: asterisk-users On Behalf Of
Jöran Vinzens
Sent: Friday, August 7, 2020 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] With ARI, is it possible to create (originate) a
call and pass both the caller id name and number?
Hi Dan,
as far as PPI and PAI Header, we use the channel Vars in order to do that. In
Latest Asterisk you
An additional follow-up question, if I need to set the P-Asserted-Identity on
the create (originate), is there a way to do this with ARI?
From: asterisk-users On Behalf Of Dan
Cropp
Sent: Friday, August 7, 2020 11:51 AM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] With ARI
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan<291>
However, when the caller id name has a space in it, I can't figure out how to
pass the name and number successfully. The following only di
a DELETE playback doesn't work during a number portion or
is this a bug?
Dan
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ext=1=mycallerid.1=mycallerid.2=ulaw=30;
Have a good day!
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Thursday, August 6, 2020 12:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is it possible to use Stasis to control both legs
of a
",
"args": [],
"channel": {
"id": "mycallerid.1",
"name": "Local/1000@mycontext-000b;1",
"state": "Down",
"caller": {
"
. Colp wrote:
> On Sun, Jul 12, 2020 at 11:37 PM Michael Maier
> wrote:
>
>> On 13.07.20 at 00:17 Joshua C. Colp wrote:
>> > On Sun, Jul 12, 2020 at 7:12 PM Dan Jenkins wrote:
>> >
>> >> Asterisk 18 will have support based on this asterisk update Ma
Asterisk 18 will have support based on this asterisk update Matt F did for
CommCon's sponsor slots
https://youtu.be/eas1csaX-wc
On Sun, 12 Jul 2020, 22:44 Steve Edwards, wrote:
> On Sun, 12 Jul 2020, Saint Michael wrote:
>
> > WORLDWIDE EMERGENCY
>
> Again?
>
> > The code below needs to be
:30 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
I have an endpoint with multiple phones registered as aor contacts.
When I attempt to originate a call it will only ring one of the phones.
Is it possible to ring multiple phones as a single endpoint. First phone to
answer wins the call a
I have an endpoint with multiple phones registered as aor contacts.
When I attempt to originate a call it will only ring one of the phones.
Is it possible to ring multiple phones as a single endpoint. First phone to
answer wins the call and all others terminated?
Again, using AMI.
Dan
son that couldnt go
out to a radio station stream for example..
On Wed, May 6, 2020 at 8:55 PM Jonathan H wrote:
> Thanks Dan - might have to scratch my head over that one for a while!
> The phrase "you make your own RTP server" has made me all twitchy ;)
>
> Jonathan
>
> O
Hi Jonathan,
I'd probably go down the external media route in the ARI now - you make
your own RTP server and provide your own RTP back to asterisk
On Sun, 3 May 2020, 13:07 Jonathan H, wrote:
> Way back in 2016 the only way to allow callers to listen in to a stream
> "at will" was to do the
it
working on most recent version of Asterisk
On Tue, Apr 28, 2020 at 11:37 AM Teijo wrote:
> Hello,
>
>
> Currently audio conference. Should upgrading Asterisk from 13 to newer
> version resolve webrtc/iOS problem?
>
>
> Best regards,
>
>
> Teijo
>
> Da
First things first, upgrade from 13 - WebRTC has moved a long a lot since
then. If you can't upgrade everything to 13 then run another asterisk
specifically for WebRTC and bridge to your other Asterisk
Is this just an audio conference?
On Sun, Apr 26, 2020 at 10:21 PM Teijo wrote:
> Hello,
>
I've heard of people having thousands of channels on a box Dovid. Not how I
would personally do it myself but if you've already got the hardware.
And think about if you can simplify the deployment by accessing the sound
files via http so you only have them in one place
On Thu, 19 Mar 2020,
Thank you Joshua.
That’s awesome!
Dan
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Monday, February 24, 2020 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can an ARI Bridge support more than 2 channels
the way a ConfBridge can
We are looking to migrate from AMI to ARI.
We currently rely heavily on ConfBridges for multiple party support.
Is it possible to add more than 2 channels?
If so, is there a limit?
Or a way to configure the limit?
Have a great day!
Dan
Thanks Joshua
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Friday, February 14, 2020 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Question on pjsip.conf and aors
On Fri, Feb 14, 2020 at 3:04 PM Dan Cropp
mailto:d...@amtelco.com
. As in [aor3] becomes [1004]
and in the endpoint change aors = aor3 to be aors = 1004
Is there a setting I'm missing to allow the endpoint named 1004 to use an auth
that doesn't have the same 1004 name?
Dan
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he channel
request hangup that it calls.
-Original Message-
From: asterisk-users On Behalf Of Dan
Cropp
Sent: Thursday, January 16, 2020 1:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] From the CLI, how can I hangup a channel name
tha
nk you for the help.
Dan
-Original Message-
From: asterisk-users On Behalf Of
Doug Lytle
Sent: Thursday, January 16, 2020 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] From the CLI, how can I hangup a channel name
that includes a space cha
trol character(s) for the CLI to interpret everything in
between as a single argument?
Dan
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ween Avaya and Asterisk. This type of transfer seems very
insecure. It's basically, Avaya able to tell us to transfer to any number they
want (without any restriction).
Have a great day!
Dan
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-users] Is it possible to record 2-4 party call audio in
stereo quality as opposed to mono?
On Friday 01 November 2019 at 22:29:28, Dan Cropp wrote:
> We have a customer who wants us to record anywhere from 2-4
> participants on a call in stereo (as opposed to mono) quality audio.
I'm as
We have a customer who wants us to record anywhere from 2-4 participants on a
call in stereo (as opposed to mono) quality audio.
Some background..
We are using asterisk 16.6.1
We are also currently using AMI/AsyncAGI and ConfBridge to bring the parties
together. I believe recording in the
Thanks Joshua.
This turned out to be my mistake.
Quiet variable was enabled on the User and needed to be disabled.
It's been at least a couple years since I wrote e-mails for my coworkers and
forgot that setting.
Have a great day!
Dan
-Original Message-
From: asterisk-users On Behalf
is working.
Did the naming for the CONFBRIDGE bridge variables changed?
Dan
From: asterisk-users On Behalf Of Dan
Cropp
Sent: Tuesday, October 22, 2019 3:54 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ConfBridge and sound prompts
We have a product that uses Asterisk via AMI.
I am
Channel: PJSIP/1003-0003
Variable: CONFBRIDGE(bridge,sound_join)
Value: en/confbridge-join
Does anyone know if the ConfBridge sound variable setting approach changed?
Dan
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We are using AsyncAGI with AMI.
On a customer box running asterisk 16.1.1, we are seeing times where asterisk
logs indicate it's started the agi:async extension.
Event: Newexten
...
Application: AGI
AppData: agi:async
It's taking 2 or more seconds before we see the
Event: AsyncAGIStart
I have
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