[asterisk-dev] who will receive the SIP response first, SIP driver or the channel?

main/channel.c So far with my testing handle_response_invite is receiving the frame data first and then the ast_read, will this always be the case or is this a race thing where either of the functions can get the data first? Thanks, Mohit Dhiman

Re: [asterisk-dev] when will be the reason for not having a channel in monitor_dial of dial.c

10 May 2021 at 18:00, Joshua C. Colp wrote: > On Mon, May 10, 2021 at 9:09 AM Mohit Dhiman > wrote: > >> Hi, >> I am having trouble understanding a piece of code in the function >> monitor_dial of the file main/dial.c >> >> who = ast_waitfor_n(cs, pos, &time

[asterisk-dev] when will be the reason for not having a channel in monitor_dial of dial.c

Hi, I am having trouble understanding a piece of code in the function monitor_dial of the file main/dial.c who = ast_waitfor_n(cs, pos, &timeout); /* If the timeout no longer exists OR *if we got no channel* it basically means the timeout was tripped, so handle it */ if (!timeout || !who) { t

Re: [asterisk-dev] External scripts for parsing the security logs

2 -0500, Sean Bright wrote: > > On 11/23/2020 4:09 AM, Mohit Dhiman wrote: > > > can anyone please recommend any existing external scripts that can > > > parse the Asterisk security logs and possibly take appropriate > > > actions like IP blocking. > > >

[asterisk-dev] External scripts for parsing the security logs

Hi, can anyone please recommend any existing external scripts that can parse the Asterisk security logs and possibly take appropriate actions like IP blocking. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.co

[asterisk-dev] impact of removing static payload from rtp_engine

Hi, I want to remove some static payload from Asterisk's RTP Engine. so if I remove the following lines from ast_rtp_engine_init of main/rtp_engine.c add_static_payload(102, ast_format_siren7, 0); add_static_payload(103, ast_format_h263p, 0); add_static_payload(104, ast_for

Re: [asterisk-dev] Segfault because RTP frame datalen negative

Hi, is there any chance that Asterisk would interpret an RTP frame as AST_FRAME_TEXT even if it is not of protocol T.140 or T.140red? On Fri, 31 Jan 2020 at 18:42, Mohit Dhiman wrote: > Thanks Joshua > > On Fri, 31 Jan 2020 at 18:18, Joshua C. Colp wrote: > >> On Fri, Jan 3

Re: [asterisk-dev] Segfault because RTP frame datalen negative

Thanks Joshua On Fri, 31 Jan 2020 at 18:18, Joshua C. Colp wrote: > On Fri, Jan 31, 2020 at 8:40 AM Mohit Dhiman > wrote: > >> turns out Asterisk-13.21 does not have the ast_rtp_interpret is there a >> similar entry point to interpret RTP packets? >> > > It

Re: [asterisk-dev] Segfault because RTP frame datalen negative

turns out Asterisk-13.21 does not have the ast_rtp_interpret is there a similar entry point to interpret RTP packets? On Fri, 31 Jan 2020 at 18:05, Mohit Dhiman wrote: > I'm using Asterisk-13.21. > I'll check out the code in ast_rtp_interpret but the problem is that I do >

Re: [asterisk-dev] Segfault because RTP frame datalen negative

r RTP? On Fri, 31 Jan 2020 at 16:51, Joshua C. Colp wrote: > On Fri, Jan 31, 2020 at 3:06 AM Mohit Dhiman > wrote: > >> Hi, >> I'm trying to debug a segfault in ast_frdup which happened because of >> the negative datalen of the ast_frame for frame type AST_FRAME_

[asterisk-dev] Segfault because RTP frame datalen negative

Hi, I'm trying to debug a segfault in ast_frdup which happened because of the negative datalen of the ast_frame for frame type AST_FRAME_TEXT. My question is that how an RTP frame in categorized as of type TEXT because I can only see two types of RTP payload in network capture (not of the time of

Re: [asterisk-dev] unable to pass default attribute_data from codec_xyz.c to res_format_attr_xyz.c

Ok, Thanks Joshua. I'll try and make it work. On Wed, 28 Aug 2019 at 17:17, Joshua C. Colp wrote: > On Wed, Aug 28, 2019, at 8:36 AM, Mohit Dhiman wrote: > > Ok, now I am a little confused here because when Asterisk initiate a > > SIP transaction (INVITE) > > and it g

Re: [asterisk-dev] unable to pass default attribute_data from codec_xyz.c to res_format_attr_xyz.c

. Colp wrote: > On Wed, Aug 28, 2019, at 8:10 AM, Mohit Dhiman wrote: > > Ok. > > could it be related to the order in which modules get loaded in > > asterisk? > > also can you please suggest some way to pass this information somehow > > from codec_xyz.c to ref

Re: [asterisk-dev] unable to pass default attribute_data from codec_xyz.c to res_format_attr_xyz.c

wrote: > On Tue, Aug 27, 2019, at 7:41 AM, Mohit Dhiman wrote: > > Hi, > > I am trying to pass default data from codec_xyz.c to > res_format_attr_xyz.c > > > > what I did: > > * main/codec_builtin.c: ast_codec_builtin_init: made a call to > > CODEC_REGISTE

[asterisk-dev] unable to pass default attribute_data from codec_xyz.c to res_format_attr_xyz.c

Hi, I am trying to pass default data from codec_xyz.c to res_format_attr_xyz.c what I did: - main/codec_builtin.c: ast_codec_builtin_init: made a call to CODEC_REGISTER_AND_CACHE(xyz) - codecs/codec_xyz.c: load_module: get the codec. xyz_codec = ast_codec_get() - codecs/codec_xyz.c

[asterisk-dev] making the .so file stand-alone such that it can be loaded by an external program

t? *if yes:* can someone please explain how to do it? *if not:* how Digium make it work independently? Thanks, Mohit Dhiman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To U

Re: [asterisk-dev] ast_data_buffer equivalent in Asterisk 13

Ok, Thanks Joshua. I'll give it a try. On Mon, 5 Aug 2019 at 18:38, Joshua C. Colp wrote: > On Mon, Aug 5, 2019, at 9:43 AM, Mohit Dhiman wrote: > > I am basically trying to use Sylvain Boily > > <https://github.com/sboily>'s implementation of res_ari_stream.c &g

Re: [asterisk-dev] ast_data_buffer equivalent in Asterisk 13

ame data received via ast_framehook_interface's event_cb into *ast_data_buffer* structure and then writing it to a websocket by continuously reading from *ast_data_buffer*. On Mon, 5 Aug 2019 at 15:05, Joshua C. Colp wrote: > On Mon, Aug 5, 2019, at 4:48 AM, Mohit Dhiman wrote: > >

[asterisk-dev] ast_data_buffer equivalent in Asterisk 13

Hi, can anyone suggest if there is an equivalent (or closest implementation) of ast_data_buffer in Asterisk 13? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE o

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

t as a mention, though it uses my AudioSocket rather than what George > is talking about, I do have a complete example of bidirectional > communication with Google TTS and speech-to-text. > > Https://GitHub.com/CyCoreSystems/asterisk-k8d-demo > > > On Thu, Aug 1, 2019, 14:4

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

towards my project. Thanks Mohit Dhiman On Thu, 1 Aug 2019 at 23:48, George Joseph wrote: > > > On Thu, Aug 1, 2019 at 12:10 PM George Joseph wrote: > >> >> >> On Thu, Aug 1, 2019 at 9:56 AM marek wrote: >> >>> is there someone who can w

Re: [asterisk-dev] Opus codec Development

Nevermind about thr SDP parameters, i didn't have the *res_format_attr_opus* module loaded. Its working fine now. On Thu, 27 Jun 2019, 12:33 pm Mohit Dhiman, wrote: > Hi Tzafrir, > I tried the open source version and compiled with some conf cahanges in > opus.h but it doesn't

Re: [asterisk-dev] Opus codec Development

> On 25/06/2019 13:47, Joshua C. Colp wrote: > > On Tue, Jun 25, 2019, at 7:44 AM, Mohit Dhiman wrote: > >> Hi, > >> Can anybody please share what is the development process of opus codec > >> in Asterisk? > >> Does Asterisk uses the fork from official xip

Re: [asterisk-dev] Opus codec Development

Thanks Joshua, Can you comment on the stability of codec_opus module and how well it can work in real time scenario considering the transcoding involved? Thanks and regards, Mohit On Tue, 25 Jun 2019, 4:17 pm Joshua C. Colp, wrote: > On Tue, Jun 25, 2019, at 7:44 AM, Mohit Dhiman wrote: >

[asterisk-dev] Opus codec Development

Hi, Can anybody please share what is the development process of opus codec in Asterisk? Does Asterisk uses the fork from official xiph opus repository for development process? Can people other than Digium contribute to Asterisk's opus codec development? How often the new opus codec module is releas

Re: [asterisk-dev] No MTU discovery and packet sizing in DTLS implementation by RTP engine (res_rtp_asterisk) which leads to IP fragmentation

Thanks Matt for the clarification. Its kind of causing problem for me, the problem is not that big though. I am really looking forward to work on this particular topic. Thanks and Regards, Mohit On Fri, 19 Apr 2019, 7:53 pm Matt Fredrickson, wrote: > On Fri, Apr 19, 2019 at 5:29 AM Mo

[asterisk-dev] No MTU discovery and packet sizing in DTLS implementation by RTP engine (res_rtp_asterisk) which leads to IP fragmentation

As per the RFC 4347 section-4.1.1 Each DTLS record MUST fit within a single datagram. In order to avoid IP fragmentation [MOGUL], DTLS implementations SHOULD determine the MTU and send records smaller than the MTU. DTLS implementations

Re: [asterisk-dev] Difference between ConfBridge and Bridge

ok, thanks Joshua. this really helps. On Thu, 27 Dec 2018 at 17:09, Joshua C. Colp wrote: > On Thu, Dec 27, 2018, at 4:30 AM, Mohit Dhiman wrote: > > Hi, > > As much as i know about Bridge application (in features.c) it uses > > masquerading to clone a channel's

[asterisk-dev] Difference between ConfBridge and Bridge

is it the same process as it is in case of Bridge? Regards, Mohit Dhiman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit:

[asterisk-dev] Bridging and Masquerades

As we only use Masquerade when we want to convert a non-pbx thread to a pbx thread. in case of two party bridging both the channels are already pbx thread (assuming both gets originated by originate cli command) then why we do Masquerade in this case also? -- __

[asterisk-dev] topic subscribers not getting removed from topic on channel cleanup

Hello, this question relates to issue ASTERISK-28197 In case of Bridge application when bridge_exec from features.c is invoked it calls the ast_channel_unref(current_dest_chan) to remove the channel reference if channel is not in the Bridge,