[asterisk-users] multiple sip trunks

[John Taylor]

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 <asterisklist at callthem.info>:
> On Fri, Dec 11, 2009 at 10:23 AM, John Taylor <john at 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 <tnelson at rockbochs.com>:
>>> ----- "John Taylor" <john at 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
>>>
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