Hi everybody,

I am trying to make up call flow diagrams for for a setup which
include ser as a sip proxy/registrar and asteriks as a voicemail
server.

Is my sequence correct?:
UA 1 send an invite to SER. SER forwards this invite to UA2. UA2
sends back a sends back a 100 trying and 180 ringing message. SER
forwards these. However UA2 doesnt answer the phone,so what happens
then?...is there a timeout message?...I know SER sends a notify
message to asterisk at some stage but im not sure of the exact
sequence or if asterisk contacts ua1 directly or through ser.
Somekind of call flow diagrams for this implementation wold be great.

Im also trying to implement this in practice. I have ser as a
registrar and asterisk set up aswell. I have modifed ser.cfg to
rewritehostport(asterisk ip:5061) when not found, however could
someone tell me what to modify in my
sip.conf,exntensions,voicemail.conf? A simple example if possible
please because all the examples I havee seen so far have pstn
forwrading implemented also which complicates things. A look at
someones working version of these would be great!

All help appreciated,
Thank you,
Aisling.



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