[SR-Users] Sip proxy for Asterisk with RFC3966 and 100Rel

[paolo.visnoviz at vipcomputers.it]

Thank you David,

 From what I understand Kamailio can work as a sip proxy and, in some 
way, I can interface it with Asterisk. Kamailio supports both 100Rel and 
RFC 3966, but my question is whether it can also be work as a "translator".

That is, if I can call from Asterisk to Kamailio and vice versa 
with/without RFC 3966 and somehow I can tell Kamailio to add (and strip) 
the missing RFC 3969 part.


Il 13/05/19 23:49, David Villasmil ha scritto:
> Someone correct me if I’m wrong, but I believe Kamailio supports both.
>
> http://www.kamailio.org/docs/modules/0.9.x/uri.html#AEN128
> https://kamailio.org/docs/modules/4.4.x/modules/siputils.html
>
> Hope that helps
>
> On Mon, 13 May 2019 at 18:32, paolo.visnoviz at vipcomputers.it 
> <mailto:paolo.visnoviz at vipcomputers.it> 
> <paolo.visnoviz at vipcomputers.it 
> <mailto:paolo.visnoviz at vipcomputers.it>> wrote:
>
>     Dear Sirs,
>
>     a telco provider give us a trunk that require 100rel and TEL RFC
>     3966.
>     Usually we use Asterisk. Now Asterisk 16 with PJSIP support
>     100Rel, but
>     not RFC3966, and they don't actually works to implement it. Asterisk
>     chan_sip, instead, support RFC3966, but not 100Rel and is not
>     actively
>     developed on by Digium or Sangoma. So I'm stuck.
>
>     Now, do you think it was possible to use Kamailio to act like a proxy
>     sip but implementing RFC3966 for incoming and outgoing calls and
>     RFC3966
>     and 100Rel for outgoing calls?
>
>     And it is possible to connect all to an Asterisk server?
>
>     Thank you in advance
>
>     Best regards.
>
>     Paolo Visnoviz
>
>
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>
> -- 
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com 
> <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
>
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