I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf
Thanks John 2009/12/11 Martin <asteriskl...@callthem.info>: > On Fri, Dec 11, 2009 at 10:23 AM, John Taylor <j...@vetsurgeon.org.uk> wrote: >> Thanks - have done that and am now trying a one out. However, I'd >> still like to know whether 1 asterisk server can support multiple >> trunks/registry entries. Does it cause problems? > yes, Asterisk does support multiple registry entries... > if it didn't ... it would be just a crippled sip endpoint > > lets say more ... Asterisk can do whatever you want it to do (within > reason and technical boundaries); > just code it in or request a feature > > Martin > >> >> Thanks >> >> John >> >> 2009/12/3 Tim Nelson <tnel...@rockbochs.com>: >>> ----- "John Taylor" <j...@vetsurgeon.org.uk> wrote: >>>> I want to use an asterisk box to provide a voip service to a number >>>> of >>>> separate companies. >>>> >>>> I have a VOIP provider who I want to trunk with. As far as I can see >>>> it there are 2 options >>>> 1. Have 1 SIP trunk to one account at the provider who gives me >>>> multiple incoming numbers; this is less than optimal as the provider >>>> does not provide the DID number in the sip header; I only get the >>>> account number. I have the option to set "called line presentation" >>>> but this will stop CLID >>>> >>>> 2. Have multiple sip trunks to multiple accounts at the provider. Is >>>> this an advisable thing to do? I notice asterisk does not handle the >>>> incoming context correctly (all incoming calls go to the last >>>> incoming >>>> context defined in sip.conf), but I can extract the account called >>>> via >>>> the EXTEN variable. >>>> >>>> I would be looking at providing around 20 companies with accounts >>>> (all >>>> very small), and would prefer option (2) to enable failover to a >>>> number they specify. >>>> >>>> Thanks for any light shed >>>> >>>> John >>>> >>> >>> Why not go with a real carrier that can send you proper DID and DNIS >>> information for each call? Rather than trying to configure/code/etc around >>> the problem with the ITSP, use an ITSP that does things correctly. There >>> are many people here on asterisk-users that can recommend a proper ITSP. If >>> you want pure business response, head over to asterisk-biz and ask there. >>> >>> Tim Nelson >>> Systems/Network Support >>> Rockbochs Inc. >>> (218)727-4332 x105 >>> >>> _______________________________________________ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> >>> 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 -- >> >> 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 -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users