Can you please post: 1. Relevant sip.conf 2. sip debug when trying to make a call?
On Thu, Sep 22, 2011 at 7:26 PM, David Björkevik <da...@dynamore.se> wrote: > Dear list, > > We are switching to a new provider for SIP-trunks. We have 20 numbers, > each defined as a separate SIP peer. > > With the old provider everything works. > > When switching to the new provider's account data, it only works as long > as I only define one of the accounts. If multiple accounts are defined, > I can only place outgoing calls on one of them, for the other(s) > authentication fails, "FORBIDDEN". > > It is almost like Asterisk is using just one of the defined passwords to > authenticate all peers on that host. > > Any input is very appreciated. > > Regards > David Björkevik, Engineer > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
