hello,
thanks Kevin, I use ast_check_hangup(chan_peer) and I can get the user CANCEL.
for now my APP work ok, I’ll test in production.
Thanks
> On 30 Jan 2020, at 16:04, i...@magnussolution.com wrote:
>
> Hi Kevin, thanks for your help
>
> I’ll to try your recommendation and send you
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 would be in ast_rtp_read in that
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 would be in ast_rtp_read in that version.
--
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
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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
> not have the access
I'm using Asterisk-13.21.
I'll check out the code in ast_rtp_interpret but the problem is that I do
not have the access to the production environment to recreate this issue.
can anybody suggest any tool to create dummy RTP payloads or some SIP
client that can generate real-time text over RTP?
On
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_TEXT.
>
> My question is that how an RTP frame in categorized as of type TEXT
> because I can only
