SearchSearch  Log in to check your private messagesLog in to check your private messages  recent posts Recent Posts
Post new topic   Reply to topic
View previous topic Printable version Log in to check your private messages View next topic
Author Message
rowellOffline



Joined: Aug 07, 2009
Posts: 5

Status: Offline
Posted: Jan 11, 2010 - 08:53 PM Reply with quote Back to top
I have a setup where I want to use Asterisk as Voicemail server. Now I always receive the message

Failed to authenticate on INVITE to '"92001" <sip:opensertrunk@domain>;tag=as5895f3a1'
-- SIP/opensertrunk-04546b90 is circuit-busy

PSTN -> Asterisk -> OpenSER

my phones are registered on OpenSER so when they are busy then they are redirected back to Asterisk for voicemail.

Any ideas?

Thanks
View user's profile Send private message
micondaOffline



Joined: Feb 02, 2007
Posts: 354
Location: Germany
Status: Offline
Posted: Jan 15, 2010 - 08:28 AM Reply with quote Back to top
You should not authenticate in asterisk the calls from kamailio (openser), trust the traffic by ip.

See here one way of integrating kamailio and astetisk:
http://www.voip-info.org/wiki/view/Real ... ilio+1.5.x
View user's profile Send private message


View previous topic Printable version Log in to check your private messages View next topic

Post new topic   Reply to topic
Forum Rules and Guidelines | About VoIP User | Privacy Policy


All logos and trademarks in this site are property of their respective owner.
Comments and posts are property of the poster, all the rest (c) 2003-2008 VoIP User Limited.

VoIP User Limited is incorporated in England and Wales under Company Number 6694577.

No part of this site may be reproduced without our prior consent.