[asterisk-users] how to read sip debug
hawat.thufir at gmail.com
Tue Jul 5 21:28:54 CDT 2016
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.
Reliably Transmitting (NAT) to 18.104.22.168:5060:
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?
70 hops max?
From: "asterisk" <sip:asterisk@<externip>>;tag=as1a7aca46
from my externip, with a hash to keep the calls straight?
easy, to telnyx
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
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
enumerating accepted replies?
no data, just "hi"
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?
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the asterisk-users