[asterisk-users] Resetting Marker Bits

[Adrian Marsh]

Hi All,

 

I'm looking for how to enable SIP Markers, or specifically, how to have
the TIME reset when a call route changes.

 

I'm debugging an issue, where a sip client we have switching to
one-way-audio, when an asterisk server fruther down the call path dials
out to the PSTN. Scenario is:

 

SIP Client ->  A*k1  -> A*k2   ->  PSTN Provider/Gradwell  -> O2  ->
Mobile

 

-         the SIP client dials on O2 mobile, call goes out to A*1.

-         A*1 Dials out to A*k2 as A*k2 is the gateway to PSTN providers
and normal office phones.

-         A*k2 dials some local Cisco phones, then on no answer plays an
audio file, so call is ANSWERED.

-         A*k2 then Dials out to gradwell, to a mobile phone number.

-         Gradwell takes the call, routes it via PSTN.

 

My problem, is that at the point where the O2 mobile accepts the call, I
get one-way audio. (SIP Client outbound, nothing inbound).

 

Tracing the RTP stream all the way back, I can see that audio makes it
all the way to the SIP Client.

However,  we notice that at the point where the O2 mobile answers, the
TIME= value of the packet jumps significantly, say from 119248 to
1518324408.

 

Talking to the sip client developer, they say that I need to enable SIP
Markers on the server (I guess A*k2), so that if the stream source
changes then the timers are reset.

Does this sound right, and if so, how do I do that ?

 

I am running an older load on A*K2 of 1.4.18, and 1.4.15svn (privately
compiled to add an extra codec) on A*k1.   I can look into upgrading
these, but the developer thinks it's just a missing config on Asterisk.

 

Thanks,

 

Adrian

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