[Asterisk-Users] sip registration problems

[Mark Spencer]

> REGISTER sip:testi.fi SIP/2.0
> Via: SIP/2.0/UDP 195.10.149.20:5062
> CSeq: 253 REGISTER
> To: "Juha Heinanen" <sip:+358333983950 at testi.fi>
> Expires: 900
> From: "Juha Heinanen" <sip:+358333983950 at testi.fi>
> Call-ID: 1697793338 at 195.10.149.20
> Content-Length: 0
> Event: registration
> Allow-Events: presence
> Contact: "Juha Heinanen" <sip:jh at 195.10.149.20:5062;transport=udp>;methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"

I believe you could emulate this with:

register => +35833983950 at testi.fi/jh

> the resolver procedure is documented in rfc 3263.  it is quite complex,
> because first an naprt lookup should be done and then an srv query.  if
> asterisk only support udp, it could skip the naptr query and just query
> the svr record of _sip._udp.domain-in-question or if it supports both
> tcp and udp and possibly also sips (tls), it could try the names in the
> order it prefers.  here is an example from the rfc:

I think the implementation of NAPTR may be coming in the near future.

Mark