Some background...
We use AMI and AsyncAGI to be able to receive events about calls (and other
Asterisk details) and control it from our application.
Works great and have about 100 sites (some newer, some older) without issues.
I was notified this morning about a customer who had something
I am working on a project that uses Asterisk ARI ExternalMedia request to
stream the RTP audio from Asterisk to an UDP/RTP receiver project.
Using slin16 format.
1) I believe I am seeing is a 12-byte header followed by 640 bytes of data. Is
this correct?
2) Is there some place I can find a
Not sure where to mention this. Very minor/trivial issue. Just wanted to let
someone know.
If you go to docs.asterisk.org and the Asterisk REST Interface (at least in
both 18 and 20 versions). Go to the Channels.
There is a list of Method and path links.
Most work, but a few do not. Not sure
I don't recall seeing the rtp and jitter entries being logged regularly at
other customer sites, so I am probably missing something obvious. This site is
running asterisk 18.17.1.
I enabled debugging to try to help track down an issue. The problem is the
debug_log file is filling rapidly with
cussion
Subject: Re: [External] [asterisk-users] Encountered a crash, what is best way
to tell if it has been fixed or now
On Wed, 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.
, 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
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
https://github.com/asterisk/asterisk/compare/18.17.1...18.19.0 to see if any of
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
isk rtp.conf stunaddr setting -
what happens if there is an outage
On Mon, Feb 6, 2023 at 6:05 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
A quick follow-up.
Looking at other customers running 18.12.1 who reported problems at the exact
same time with AWS issue described below.
We are seein
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
Over the weekend, we had several customers running at AWS. AWS had an outage
during this time.
This customer is running Asterisk 16.23.0 (which has the STUN timeout crash
fix).
>From what I have been told, other customers are running newer Asterisk 18.12.1
>but encountered similar issues. (I
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
We have used AMI for many years and I'm in the process of migrating to ARI.
My understanding is the call should be handed off to Stasis for the ARI
application to control it.
I was playing around with things and discovered the ARI hangup (DELETE
/channels/{channelId}) allowed me to hangup
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
A couple years ago, I know I had ARI externalMedia working. Trying to figure
out what I'm doing wrong today.
https://wiki.asterisk.org/wiki/display/AST/External+Media+and+ARI
My ari.conf
[general]
enabled = yes
pretty = no
allowed_origins = *
[MyApp]
type = user
read_only = no
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:
introduced a behavior where PJSIP is unable to send a response to
OPTIONS (seems to resolve after anywhere a period of time)
On Wed, Dec 7, 2022 at 11:34 AM Joshua C. Colp
mailto:jc...@sangoma.com>> wrote:
On Wed, Dec 7, 2022 at 11:26 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
On
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
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
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk
, 2022 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] Question on Originate with EarlyMedia
On Thu, Sep 1, 2022 at 1:32 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Using AMI, we send an Originate with EarlyMedia: true s
Using AMI, we send an Originate with EarlyMedia: true setting
If the other end sends a 183, Asterisk
When the 183 is received, Asterisk indicates the ChannelState: 6 and
ChannelStateDesc: Up values.
All is fine up to this point.
It may take the caller several seconds before the called party
at fix that and a few other bugs.
On Tue, Aug 23, 2022 at 2:47 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Running into a problem when retrieving the profile_precedence in the
extensions.conf
Creating a very basic geolocation.conf to allow passing through geolocation
values
Running into a problem when retrieving the profile_precedence in the
extensions.conf
Creating a very basic geolocation.conf to allow passing through geolocation
values for outbound.
[discard_config]
type = profile
profile_precedence = discard_config
[discard_incoming]
type = profile
Sent: Tuesday, August 16, 2022 7:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] [External] Geo location 18.14.0-rc1
question
On Mon, Aug 15, 2022 at 1:59 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Thank you George.
Goo
@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
Thank you George.
Good idea on the passthrough profile.
Is there a way to set the GEOLOC_PROFILE values from the AMI Originate command?
I tried the following, but it doesn’t like the GEOLOC_PROFILE values in the
variable parameter. If there is a way to do this, the passthrough would solve
Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] [External] [External] [External]
[External] Geo location 18.14.0-rc1 question
On Fri, Aug 12, 2022 at 10:01 AM George Joseph
mailto:gjos...@sangoma.com>> wrote:
On Thu, Aug 11, 2022 at 8:43 AM Dan
an into some
strange issues.
On Tue, Aug 2, 2022 at 1:43 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Thank you George
From: asterisk-users
mailto:asterisk-users-boun...@lists.digium.com>>
On Behalf Of George Joseph
Sent: Tuesday, August 2, 2022 2:40 PM
To: Asterisk Users Maili
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
I believe I have everything configured correctly, but Asterisk is complaining
about my configuration
It is complaining about confidence settings.
>From the Asterisk Geolocation Implementation Wiki, I believe I have this set
>correctly.
Sub-parameters:
* value: A percentage indicating
From: asterisk-users On Behalf Of
George Joseph
Sent: Wednesday, July 27, 2022 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [External] [asterisk-users] Question about the Geo Location
support being added
On Wed, Jul 27, 2022 at 11:02 AM Dan Cropp
mailto:d
Looking at the Asterisk wiki
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Geolocation+Implementation
I see the dial plan support the GeolocProfileCreate and there is support for
GEOLOC_PROFILE settings to be set on the dial plan.
We currently use AMI Originate support. We may have
I believe the answer #2 depends on the user options for each participant.
If all participants have user options with wait for marked set to true there
will be no conference/recording until at least one marked user joins.
If any participants have user options with wait for marked set to false,
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
Thank you Joshua.
I will read both links you provided.
As mentioned in another response. If what our customer requires doesn’t fit
well with Asterisk development, we believe we can do this work with Kamailio
front ending the calls. Something customer is already requiring us to do for
high
.
Från:
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
mailto:asterisk-users-boun...@lists.digium.com>>
För Dan Cropp
Skickat: den 23 maj 2022 22:01
Till: Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@list
Out of curiosity, is there any documentation on what is planned for the
Geolocation/E911?
Is the plan for Asterisk to expose the SIP body and leave it to dial plan, ARI,
AMI to process the data?
For example, mime pidf+xml section?
Or is there a different approach being worked on (or planned to
presence question
On Fri, May 20, 2022 at 1:59 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Thank you Joshua.
Any guess on a timeframe for geolocation/E911 support being part of an Asterisk
version?
A month or two?
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check
] 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
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
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
Hi Joshua,
Thank you for helping me diagnose this.
Interesting that they are the exact same between versions.
File sizes are slightly different between the two when I compile them. 18.12.0
would have any new default configurations that were not part of 18.11.2, such
as aeap. Some codecs also
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,
)
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,
] Question on ExternalMedia and the codec
On Tue, Oct 12, 2021 at 2:54 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
We tell asterisk to use the slin format for ExternalMedia. However, the
unicast channel is selecting ulaw formatand the RTP data is indicating it’s
ulaw format.
Anyone kn
We tell asterisk to use the slin format for ExternalMedia. However, the
unicast channel is selecting ulaw formatand the RTP data is indicating it's
ulaw format.
Anyone know why ulaw format would be on chosen?
[10/12 16:13:39.396] DEBUG[1665] http.c: HTTP Request URI is
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 is
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
Running Asterisk 16.17.0
We combine calls into a ConfBridge using AMI with AsynAGI. Executing actions
to ConfBridge channels into the same ConfBridge.
It's a very large and busy system, so there are dozens of these that may happen
during a second (different channels), so thousands in an hour.
We have an extremely busy/large customer. They run fine most of the time, but
periodically asterisk will output FRACK refcount related messages. It doesn't
seem to be related to the volume, because it's not breaking during their peak
times.
When this happens, the system becomes unstable and
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
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 cores.
Things work fine most of the time.
They
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 =
Has anyone used Audio Sockets with Amazon Transcribe?
I'm still very new to the Audio Socket and have only just started looking at
Amazon's Transcribe documentation so there may be something I am missing.
I'm looking for live transcription of the call as opposed to post call
transcription.
Is
Running Asterisk 16.17.0
We have an interesting scenario where we see Asterisk CPU usage spike to the
point the entire system is maxed out.
There is a specific scenario where we have two ConfBridges and they are
connected via a local channel. Everything is fine here.
Callers <-> ConfBridge A
One of my co-workers just migrated from a Samsung phone to a Pixel 5 phone.
An app on the phone dials into our asterisk.
He has the same app installed on both and can move the SIM card between them.
Call is answered and a prompt plays to collect digits.
The app dials a number followed by a pound
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
I'm attempting to run a test of the ARI recording of audio from the channel.
When I send the record command, it's failing.
curl -v -u asterisk:asterisk -X POST
"http://locahost:8088/ari/channels/mychanntest.1/record?name=mytest=WAV=300=3;
[08/11 09:14:13.290] WARNING[23806]:
- Non-Commercial Discussion
Subject: Re: [asterisk-users] ARI Stop Playback
On Thu, Aug 6, 2020 at 7:28 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a
number is being played?
Here is a test I am running.
088/ari/channels/newChannelId;<http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world>
--data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,
"PJSIP_HEADER(add,P-Asserted-Identity":"fooba
,PAI) as you would do it in Dialplan.
https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/
BR
Jöran
On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
An additional follow-up question, if I need to set the P-Asserted-Identity on
the create (ori
can set Channel vars within the create command in the Body.
Just set the PJSIP(add,PAI) as you would do it in Dialplan.
https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/
BR
Jöran
On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
An additional
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
splays asterisk
for the number and Dan for the name
curl -v -u asterisk:asterisk -X POST
http://asterisk:astersk@localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003@1003=hello-world=1000=mycontext=1=mycallerid.1=ulaw=30=Dan
Cropp<291>
Here is an example of how we do this with AMI
Should the ARI DELETE /playback/{playbackId} be able to stop a playback when a
number is being played?
Here is a test I am running. I am playing multiple portions (sounds and
numbers).
curl -v -u asterisk:asterisk -X POST
Local channel created using ARI?
On Thu, Aug 6, 2020 at 1:52 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
I understand how to control the first local channel, but an having trouble
getting the second local channel to enter stasis.
I setup have the following extensions.conf to handle 1000 (ba
I understand how to control the first local channel, but an having trouble
getting the second local channel to enter stasis.
I setup have the following extensions.conf to handle 1000 (basically had it
setup so if first stasis not there try second, but believe second channel never
processes the
: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
--
?
On Mon, Feb 24, 2020 at 8:07 PM Dan Cropp
mailto:d...@amtelco.com>> wrote:
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
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
I have the following configuration...
[aor3]
type = aor
max_contacts = 1
remove_existing = yes
[auth3]
type = auth
username = 1004
password = SuperSecretProbation
[1004]
type = endpoint
context = IS
transport = transport1
auth = auth3
aors = aor3
accountcode = 3
dtmf_mode = rfc4733
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
Thanks Doug.
Turns out if using hangup request does not work with the escaped character
CLI> hangup request PJSIP/1003\ a-0007
Usage: channel request hangup |
Request that a channel be hung up. The hangup takes effect
the next time the driver reads or writes from the channel.
I have a customer who named their endpoint to include a space (example, 1003 a)
>From the CLI, I want to hangup a channel on this endpoint
>From core show channels concise, I see the channel name includes the space
PJSIP/1003 a-0002
I realize the space is interpreted as an argument
We have a customer using Avaya. Currently, they are using chan_sip. We are
working to migrate them to PJSIP.
I have not been filled in on the exact scenario. I suspect they have some auto
forward feature on the number. Rather than their Avaya transferring
internally, they tell Asterisk to
-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
We have a product that uses Asterisk via AMI.
I am relatively certain we used to be able to play prompts by actions like the
following to make asterisk play the confbridge-join prompt when a new user
joins the confbridge.
However, that doesn't seem to work now.
Action: SetVar
ActionID: C58
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
always do a search in the confbridge
files for the verbose message and set the verbose level to whatever number that
message specifies.
On Mon, Sep 16, 2019 at 9:44 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
I’m trying to track down a CPU spike we are seeing in a system.
We have tracke
I'm trying to track down a CPU spike we are seeing in a system.
We have tracked down the spike to a single CPU and TID using that CPU.
Indications are that it's asterisk running this TID.
I'm trying to figure out what asterisk is doing on this thread around that
time, but haven't been able to
a second CPU core jump to 80%.
Anyone have an idea why a single CPU core would spike while the others remain
low?
Have a great day!
Dan
-Original Message-
From: asterisk-users On Behalf Of Dan
Cropp
Sent: Wednesday, August 21, 2019 12:16 PM
To: Asterisk Users Mailing List - Non
@lists.digium.com
Subject: Re: [asterisk-users] ptime
On Tue, Sep 3, 2019, at 10:52 AM, Dan Cropp wrote:
>
> We have a customer with a system rejecting calls from Asterisk. It’s
> indicating the ptime is 60, but the system admin is saying they only
> support 20.
>
>
> They
We have a customer with a system rejecting calls from Asterisk. It's
indicating the ptime is 60, but the system admin is saying they only support 20.
They are running asterisk 16.2.1 and using chan_sip
Is there a way to specify this with chan_sip?
Also, for my own curiosity, is there a way to
Thank you Joshua
-Original Message-
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Wednesday, August 28, 2019 9:15 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Amazon AWS question
On Wed, Aug 28, 2019, at 11:09 AM, Dan Cropp wrote:
> Thank you Jos
, at 2:00 PM, Dan Cropp wrote:
> Thank you Joshua.
>
> Out of curiosity, what is the maximum capacity you have heard for
> simultaneous ConfBridges in a single box? (Looking at 3-4 channels
> per
> ConfBridge) with recording.
I don't really remember any specific values. 100? 200?
Thank you Joshua.
Out of curiosity, what is the maximum capacity you have heard for simultaneous
ConfBridges in a single box? (Looking at 3-4 channels per ConfBridge) with
recording.
Dan
-Original Message-
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Wednesday, August
Thanks Doug.
We are running Asterisk 16.3.0 so I think we're on a pretty good version for
the timing.
We have Asterisk running on ESXi here and it's running at several customer
sites in various VM environments.
Ironically, none of them have the latency sensitivity set to high. This is
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