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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20160705/84189e08/attachment.html>