[asterisk-users] Force Jitter Buffer for SIP to SIP calls

[Thermal Wetland]

We have a customer on a wireless connection that has very bad jitter. They
can hear people fine, but people have a very hard time hearing them. They
are connected via a SPA-2102.

It is a SIP client going to a SIP trunk.

Something like this in sip.conf [general] would be in effect for all SIP
clients:
jbenable = yes
jbmaxsize = 150
jbresyncthreshold = 1000
jbimpl = fixed
jblog = yes
I only want to enable the jitter buffer for the end points having the
trouble.

Reading the docs, it seems that the jitter buffer is only used when the end
point is connected to an app like voicemail.

-- 
-Thermal
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