Ashling O'Driscoll
2005-Jan-28 11:42 UTC
[Asterisk-Users] asterisk call flow diagrams for ser voicemail combo
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. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.