Re: [asterisk-dev] Avoid audio loss in ChanSpy (ref: ASTERISK-25866)

Le 01/04/2016 18:33, Jean Aunis a écrit : Le 01/04/2016 14:00, Joshua Colp a écrit : I think this is fine, provided there is still some high ceiling value. Allowing a queue to potentially grow out of control would be bad. I think if we were able to switch things to using a timer instead we

Re: [asterisk-dev] Avoid audio loss in ChanSpy (ref: ASTERISK-25866)

Le 01/04/2016 14:00, Joshua Colp a écrit : I think this is fine, provided there is still some high ceiling value. Allowing a queue to potentially grow out of control would be bad. I think if we were able to switch things to using a timer instead we could actually get rid of this

Re: [asterisk-dev] Avoid audio loss in ChanSpy (ref: ASTERISK-25866)

Le 01/04/2016 14:00, Joshua Colp a écrit : Jean Aunis wrote: Hello, As described in the issue ASTERISK-25866 <https://issues.asterisk.org/jira/browse/ASTERISK-25866>, it appears that ChanSpy is randomly loosing audio frames, because it sets the flags AST_AUDIOHOOK_TRIGGE

[asterisk-dev] Avoid audio loss in ChanSpy (ref: ASTERISK-25866)

le for many ChanSpy users. What do you think of the idea ? Best regards, Jean Aunis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

[asterisk-dev] Mistake in review 2520 (ChanSpy modification)

the channel, when it should also listen to the audio coming into the channel. Sorry for that, I did not test this case properly before. Should I submit a new patch on gerrit to fix it ? Best regards Jean Aunis -- _ -- Bandwidth

[asterisk-dev] Questions about Message/ast_msg_queue

ch SIP message ? 2- is it technically possible (and even desirable) to change this behaviour ? Best regards, Jean Aunis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing

[asterisk-dev] Adding an audio format in Asterisk

then : - firstly, is this really the only thing to do, or am I missing something ? - secondly, is there a more "pluggable" way to do this ? Maybe with a shared object which would be loaded on startup ? Best regards

Re: [asterisk-dev] Adding an audio format in Asterisk

Le 24/03/2017 à 23:01, Matt Fredrickson a écrit : On Fri, Mar 24, 2017 at 7:43 AM, Jean Aunis <jean.au...@prescom.fr> wrote: Hello, I need to add support for a new audio format in Asterisk. It will be actually a very limited support : I just need to take the audio from one side and tr

[asterisk-dev] Adding a call preemption feature

otherwise         * PREEMPTED_CHANNEL: the name of the channel that was hanged up in case of success, empty otherwise By the way, would this feature have a chance to be merged upstream ? Regards Jean Aunis -- _ -- Bandwidth and

Re: [asterisk-dev] Adding a call preemption feature

Le 13/11/2017 à 23:17, Phil Mickelson a écrit : Jean, You should know that I wrote something very similar to what you are asking for.  Slightly different reasons and I use ARI which makes it very easy.  However, the result is the same. The inbound call gets terminated. I also considered

Re: [asterisk-dev] Adding a call preemption feature

Le 15/11/2017 à 22:47, Joshua Colp a écrit : On Wed, Nov 15, 2017, at 05:45 PM, Jean Aunis wrote: Le 13/11/2017 à 23:17, Phil Mickelson a écrit : Jean, You should know that I wrote something very similar to what you are asking for.  Slightly different reasons and I use ARI which makes it very

Re: [asterisk-dev] Adding a call preemption feature

Le 13/11/2017 à 17:58, Steve Edwards a écrit : On Mon, 13 Nov 2017, James Finstrom wrote: Generally the idea of arbitrarily killing calls seems awful, even if the behavior is expected. Yeah so john we need to . RING John is confused, your brain has to reset because whatever was happening

[asterisk-dev] Unused channel tech ast_kill_tech ?

registered. Am I missing something obvious, or could this piece of code be safely removed ? Regards -- Jean AUNIS Ingénieur R R engineer Tel : +33 1 30 85 90 22 Standard: +33 1 30 85 55 55    Rue de Broglie    22300 LANNION    FRANCE www.prescom.fr <http://www.prescom.fr/> /&

[asterisk-dev] ARI events order

indication that the resources associated to the channel can be freed. Regards -- Jean AUNIS Ingénieur R R engineer Tel : +33 1 30 85 90 22 Standard: +33 1 30 85 55 55    Rue de Broglie    22300 LANNION    FRANCE www.prescom.fr <http://www.prescom.fr/> /"Les information

Re: [asterisk-dev] ARI events order

12:22 PM Jean Aunis mailto:jean.au...@prescom.fr>> wrote: Hello, It looks like the ARI events ordering during channel destruction is not deterministic. I noticed this for ChannelLeftBridge and ChannelDestroyed events : given a channel is in a bridge a

Re: [asterisk-dev] Regarding realtime audio streaming from mixmonitor

this with the ChanSpy application, combined with a UnicastRTP channel. Regards Jean Aunis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

[asterisk-dev] Manipulating Connected Line with ARI

. Although feasible, this is a quite complicated mechanism. I think doing the same thing purely with ARI, without the burden of writing dialplan, would be interesting. Any thoughts on that ? Regards, Jean Aunis -- _ -- Bandwidth

Re: [asterisk-dev] Audio to/from Asterisk

It may not be suitable for your use case, but you could instantiate a UnicastRTP channel. It will allocate an RTP port and store it into a channel variable. Jean Le 22/07/2019 à 10:01, Dan Jenkins a écrit : Also coming back to this with some real-life case issues I'm currently facing and why

Re: [asterisk-dev] ARI Text messaging : inconsistencies in the API ?

Le 25/02/2020 à 19:09, Kevin Harwell a écrit : I could never get (2). When trying to send variables in the TextMessageReceived event I would get a validation error unless they are formatted like (3). (3) is the currently declared documented way to to it. As such any other way breaks the

[asterisk-dev] ARI Text messaging : inconsistencies in the API ?

riable field to look like the following: ... "variables": [{ "key": "My-Custom-Header", "value": "the_value" }, { "key": "Another-header", "value": "another_value" }] I personally think formats (1) and (3) both mak

Re: [asterisk-dev] No Bridge Created Event published

Hi, When I have to deal with bridges in ARI I explicitly subscribe the application to the newly created bridge. If I remember well this is because I don't get the BridgeDestroyed event otherwise. The code (NodeJS) looks like something like this : bridge.create().then( () => {    

Re: [asterisk-dev] Advanced Codec Negotiation: Need info and uses cases

I agree with Michael's principles. I would just add one question, for which I don't have a proper answer : how do we handle codecs with different configuration parameters ? Example : Alice's endpoint is configured with the codec AMR-WB, with the parameter "octet-align" set to 1. Alice calls

Re: [asterisk-dev] Looking to input on a feature I would like to write...in depth Transfer (REFER) failure reasons

Le 08/01/2021 à 15:40, Dan Cropp a écrit : Before I submit a feature request and take ownership of it, trying to gather some feedback. I’m looking to write code for an additional feature in asterisk. Currently, when performing a Transfer (REFER), the channel variable TRANSFERSTATUS only

Re: [asterisk-dev] PJSIP messaging : specifying the user portion of the request-URI

Le 15/06/2021 à 15:28, George Joseph a écrit : [2021-06-15 10:42:49.885] WARNING[5163]: res_pjsip_messaging.c:247 insert_user_in_contact_uri: Dest: 'PJSIP/3200@linphone' MSG SEND FAIL: There's already a username in endpoint linphone's contact URI 'sip:linphone@127.0.0.1:5063

Re: [asterisk-dev] PJSIP messaging : specifying the user portion of the request-URI

Le 16/06/2021 à 16:32, George Joseph a écrit : On Wed, Jun 16, 2021 at 6:27 AM George Joseph > wrote: Changes up in gerrit... https://gerrit.asterisk.org/c/asterisk/+/16068 Thanks. I've just tested it, with both

Re: [asterisk-dev] PJSIP messaging : specifying the user portion of the request-URI

Le 09/06/2021 à 16:18, George Joseph a écrit : Hi Guys, The change for allowing a dialstring-like destination in MessageSend (pjsip only) is now committed in the 16, 18 and master branches.  You can now use MessageSend(pjsip:PJSIP/@) and the request URI will be composed of the endpoint's

Re: [asterisk-dev] PJSIP messaging : specifying the user portion of the request-URI

Le 04/05/2021 à 20:30, George Joseph a écrit : I think the easiest way to solve this is to just allow MessageSend take the same format as a simple dial string like so MessageSend(pjsip:PJSIP/8005551212@provider) "pjsip:" routes the request to the pjsip channel driver. "PJSIP/" tells the

[asterisk-dev] PJSIP messaging : specifying the user portion of the request-URI

Hi, I'm trying to send a SIP MESSAGE to a PJSIP endpoint, while specifying a destination number (that is, the "user" portion of the request URI in sip:u...@domain.com). Currently, this is only possible by specifying the full request URI. For example, someone could use: >

Re: [asterisk-dev] PJSIP Video

Le 13/07/2021 à 06:11, John T. Bittner a écrit : Hello, Anyone know how to force Asterisk into a video call even if the source invite doesn’t show video support. ?. I ask because I am dealing with a bunch of automation wall panels that have inbound video features but no camera. Inbound

Re: [asterisk-dev] Question regarding native bridge's framehook

Le 06/10/2021 à 14:35, Joshua C. Colp a écrit : On Wed, Oct 6, 2021 at 6:54 AM Jean Aunis <mailto:jean.au...@prescom.fr>> wrote: Hello, I have a question regarding the use of a framehook in bridge_native_rtp. The framehook is used to redirect RTP streams when a hol

[asterisk-dev] Question regarding native bridge's framehook

Hello, I have a question regarding the use of a framehook in bridge_native_rtp. The framehook is used to redirect RTP streams when a hold, unhold or RTP update is detected. But what's the reason for doing this in a framehook ? Why not doing it in the native_rtp_bridge_write callback ? Doing

Re: [asterisk-dev] Handling blind transfers with ARI

Le 15/06/2022 à 16:33, Joshua C. Colp a écrit : [snip] Based on the test coverage for this[1] you aren't really expected to manipulate the bridge yourself. The ";1" side of the Local channel is supposed to automatically go into your ARI application and take the transferer channel place in

[asterisk-dev] Handling blind transfers with ARI

Hello, I had no answer to this question on asterisk-app-dev, so I'm trying here. 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

[asterisk-dev] Adding support for MKI in res_srtp

Hello, I'm working on a patch to add support for MKI in res_srtp. For those who may not be familiar with the subject, MKI is a re-keying mechanism for SRTP which involves appending a Master Key Identifier (MKI) inside each SRTP packet in order to change the master key in use. Master key and

Re: [asterisk-dev] Adding support for MKI in res_srtp

Le 06/10/2022 à 12:35, Joshua C. Colp a écrit : On Thu, Oct 6, 2022 at 7:27 AM Jean Aunis wrote: [snip] From a general perspective it seems sane, but that's without any further knowledge of MKI. Thanks for the reply Joshua. I'll start this way and will provide the patch once it's