[asterisk-users] pjsip: Inbound calls: selecting the correct trunk with one provider and different numbers

Michael Maier m1278468 at mailbox.org
Fri Jun 9 13:44:36 CDT 2017

On 06/08/2017 at 10:22 PM Michael Maier wrote:
> Hello Joshua,
> thank you very much for your extremely quick answer! I really appreciate
> your work and your friendly and your patient support!
> On 06/07/2017 at 10:33 PM, Joshua Colp wrote:
>> On Wed, Jun 7, 2017, at 05:28 PM, Michael Maier wrote:
>>> Hello!
>>> I've got a problem to select the correct trunk if there is one provider
>>> and different numbers with different configurations for this same
>>> provider.
>>> Example:
>>> trunk-prov1-2345
>>> trunk-prov1-2346
>>> trunk-prov1-2347
>>> Each trunk registers an own number (at the same provider) and provides
>>> own configuration: they have different allowed codecs e.g..
>>> What I'm experiencing now, is, that each incoming call is provided by
>>> trunk-prov1-2346, no matter which number has been dialed.
>>> The problem isn't the routing (this is done on base of the correct DID),
>>> but the problem is, that wrong codices are used if the wrong trunk is
>>> selected.
>>> Is this a problem of asterisk or is it caused by the provider, which
>>> always addresses the same "trunk" regardless which number has been
>>> called?
>> Asterisk is the one who associates an incoming message with an endpoint.
>> In the case of providers you can use IP based matching - which would
>> behave as you see, only one can be matched. The second option is the
>> line option[1] which may or may not work (it depends on the behavior of
>> the provider). If it works then the right endpoint would be chosen. Out
>> of those two options there's nothing else applicable built in to match.
>> [1]
>> http://blogs.asterisk.org/2016/01/27/the-pjsip-outbound-registration-line-option/
> Unfortunately Deutsche Telekom doesn't support this solution :-(.

Further investigation showed, that Telekom provides the line info in the
Request Line (as seen by Wireshark):

Request-Line: INVITE sip:+49xxxx at;line=azpreyb SIP/2.0

You can't find it if you expect it in contact header - or do you expect
it in the Request-Line?


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