Caller doesn't hear ringing in ear when using append_branch and serial_branches/next_branches
I have the following issue.
I have a customer with phone number 9012732009 and if that number is called then I give the option for other numbers to be called first depending on how the Q Values are set up. Here are the numbers that are used
90121X8X63 <-- A number that has to be reached via PSTN
9013349019 <-- An internal number that is configured on OpenSIPS
9012732009 <-- The main number of the user
So user 9012732009 can figure the above numbers so that one is called first and if the call times out then it will call the next number depending on how the Q Value is set. The issue I am seeing is that if the number 90121X8X63 is called first and the times out and calls the next two internal numbers then the person who called 9012732009 will hear ringing in the ear when 90121X8X63 is being called but won't hear ringing when the next two numbers are being called. Yet if 9013349019 or 9012732009 picks up then two-way audio is fine. So I am not sure why the Caller can't hear ringing when those internal numbers are called. Here are the Q Values so you see how the numbers are called
90121X8X63 Q = 90 Called First If he Picks up there is audio
9013349019 Q = 50 Called Second If he Picks up there is audio (But whilst the callee's phone is ringing the caller doesn't hear the caller ringing sound in the ear)
9012732009 Q = 40 Called Last If he Picks up there is audio (But whilst the callee's phone is ringing the caller doesn't hear the caller ringing sound in the ear)
Here is the siptrace during the call
Re: Caller doesn't hear ringing in ear when using append_branch and serial_branches/next_branches
This post has NOT been accepted by the mailing list yet.
The only difference i see in your SIP trace is that for first call your media server sends 183 Early Media, while for second call (after 1st times out) your media server sends 180 Ringing instead of 183.
The difference between the two sip responses is that in case of 183 Early Media, remote ring back tone is sent by callee endpoint in form of inbound audio. While in case of 180 Ringing, there is no remote ring back tone and the caller endpoint suppose to play some local ring back tone to give audio feedback to the caller.
Therefore, it appears that your caller endpoint does not support local ring back tone, that's why you don't get any ring back.
For workaround, you can force your media server (if a media server is really involved), to send 183 early media instead of 180 ringing reply for each invite.
In this example the Caller first calls a user local to OpenSIPS. Then the
call is forked and a user out on the PSTN network is called. Finally
another fork occurs and a local user is called.
local user 9012732009 Q = 90 Called First Caller hears ringing
PSTN user 90121X8X63 Q = 50 Called Second Caller hears ringing
local user 9013349019 Q = 40 Called Last Caller hears ringing
So you would think that on the last call the Caller would not hear ringing
in his ear but he does.
So my first example had the following
Callee1 sends a 183
Callee2 sends a 180 <---- No ringing
My second example had the following
Callee1 sends a 180
Callee2 sends a 183
Callee3 sends a 180 <---- Caller can hear ring in ear just fine
So on my second scenario is it because the first callee sends a 180 that
the third callee sending a 180 doesn't mess things up??? Perhaps this
isn't a bug and we can move this conversation to the mailing list, but I
figure OpenSIPS would need a way to fix this or else other people will run
into this say issue when appending branches and calling out to PSTNs that
send back 180 when the second callee sends a 183.
I tested the first scenario with Blink being the Caller and Blink was not
able to hear ringing in the ear when the second callee was called.