VoIP Mailing List Archives

[hawat.thufir at gmail.com]

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

[hawat.thufir at gmail.com]

This is interesting:

"Note that the To and From header fields are not reversed in the response message as one might expect them to be. This is because the To and From header fields in SIP are defined to indicate the direction of the request, not the direction of the message. "  -Cisco


so, when I'm receiving an inbound call, the direction would be telnyx first, then me.  Regardless of whether the ?message? is from me or the provider.



-Thufir

[mefhigoseth at gmail.com]

Hi Thufir,

The analysis of a SIP Debug depends on what the problem to be solved.

If you experience problems with inbound calls from a SIP trunk or
provider, you can type in Asterisk cli 'core set debug 3' and then
'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP
provider or from where it is supposed to come call.

Then you make a test call, and look in full log an INVITE message
(note that you analize an OPTION message in your mail, but I think
that this not help in this case).

After the incoming INVITE message from your SIP provider, you can
follow the rest of the Asterisk logic and look for the reason why
Asterisk is denying that call.

Hope this help you.

--
GnuPG Key ID: 0x39BCA9D8
https://www.github.com/mefhigoseth
...:::[ God Rulz ! ]:::...

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk.org at sedwar...]

On Wed, 6 Jul 2016, Victor Villarreal wrote:

[isrlgb at gmail.com]

Another nice sip packet is sngrep
Shows realtime the sip flows
But i think you have to chk the asterisk answer in the dialplan logic to chk what context its hitting etc. בתאריך 6 ביולי 2016 10:05 PM,‏ "Steve Edwards" <asterisk.org@sedwards.com (asterisk.org@sedwards.com)> כתב: