[asterisk-users] how to read sip debug

[Thufir]

Generally, what am I looking for when turning SIP debug on?  More
specifically, the provider says that I'm returning a 404 when they try to
call me.  Now, I had inbound working, literally, the other day.  Outbound
works fine.  I "may" have broken it either through Asterisk config or the
providers portal with settings.  Ok, I broke it -- not sure how.

comments interspersed:

mordor*CLI>
Reliably Transmitting (NAT) to 192.76.120.10:5060:

I think/infer/assume that this is the IP address for telnyx SIP servers

OPTIONS sip:sip.telnyx.com SIP/2.0

What does OPTIONS mean?

Via: SIP/2.0/UDP <externip>:5060;branch=z9hG4bK28142189;rport

rport relates to NAT?  The message is via SIP UPD from my externip ....
what is branch?

Max-Forwards: 70

70 hops max?

From: "asterisk" <sip:asterisk@<externip>>;tag=as1a7aca46

from my externip, with a hash to keep the calls straight?

To: <sip:sip.telnyx.com>

easy, to telnyx

Contact: <sip:asterisk@<externip>:5060>

from me

Call-ID: 6fce72627f253b7f2e15dac713b52392@<externip>:5060

another hashcode, Call-ID ?

CSeq: 102 OPTIONS

?

User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3

easy enough, my system

Date: Wed, 06 Jul 2016 02:17:12 GMT

easy, date

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE

enumerating accepted replies?

Supported: replaces

?

Content-Length: 0

no data, just "hi"


---
mordor*CLI>


If I see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions in
a SIP trace, that's relatively clear.  But what am I looking for with
regards to receiving calls?


thanks,


Thufir
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