[Asterisk-Dev] chan_sip

[Nickolay Shestakov]

Hi, all
I'm the beginner in Asterisk. Sorry, if my questions are already discussed.
The first questions about chan_sip:

1.	When * registers with SIP provider, I see the following messages:
REGISTER
401 Unauthorized
REGISTER
200 OK
ACK (From * ) - It seems to me, that this ACK is superfluous. So, may be,
just delete  transmit_request(p, "ACK", 0) after receiving 200 OK ?

And I noticed, that the reactions of SIP providers are different (after this
ACK). FWD sends 200 OK, but my SIP provider (www.voipexchange.ru) returns
another ACK to me.

2.	Registration with SIP provider is ok. But, when * sends INVITE, my SIP
privider returns 401 Unauthorized. And, of course, after that * crashes with
segmentation fault, because, in function handle_response
	case 401:
		do_register_auth...

May be, the matter is, that in RFC 3261:
"The branch parameter value MUST be unique across space and time for
   all requests sent by the UA"
But, I noticed, that branch is different when * registers and when it sends
INVITE. I didn't dig deeper, just deleted parameter branch from via (ata-186
works without it, and now it works (excluding, that now my provider returns
415 No media - I don't know  why.. :)


Regards,
Nickolay
npshe at mail.ru