[asterisk-dev] Operator intercept from SIP service providers

[Wai Wu]

Hello developers,

Through working with a few sip providers recent. I found that in many
occasions, the service providers are not sending the relevant message
(sip 408). Instead, the SIT tone and the announcement are played. This
could take up to 60 seconds. For the particular application we are
doing, this hurts us very badly. I dig into the code, and realise
dsp_call_progress was not called from chan_sip.c. My question is. If SIT
tone detected desired, would async_wait in pbx.c be a good place to add
the code as we are using originate to initiate the calls. What is your
take?