asterisk users - Jul 2016 - how to read sip debug

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 ! ]:::...

On Wed, 6 Jul 2016, Victor Villarreal wrote:
> 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.
Another path to enlightenment is to use tcpdump to capture the packets to 
a file and then use wireshark.

Wireshark has a 'Telephony' menu and a 'SIP Flows' menu item
where it will
list all of the SIP packets. You can sort by column to help locate the 
packet of interest.

Once found, you can click on 'Flow Sequence' and it will pop up a window
showing the 'dialog ladder' that includes that packet. As you click on 
each packet in the flow, the main wireshark window will re-position to 
that packet so you can examine it in detail.

Also on the 'SIP Flows' window is a 'Play Streams' button. Kind
of scary
how easy this is...

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
             https://www.linkedin.com/in/steve-edwards-4244281

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.
?????? 6 ????? 2016 10:05 PM,? "Steve Edwards" <asterisk.org at
sedwards.com>
???:
> On Wed, 6 Jul 2016, Victor Villarreal wrote:
>
> 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.
>>
>
> Another path to enlightenment is to use tcpdump to capture the packets to
> a file and then use wireshark.
>
> Wireshark has a 'Telephony' menu and a 'SIP Flows' menu
item where it will
> list all of the SIP packets. You can sort by column to help locate the
> packet of interest.
>
> Once found, you can click on 'Flow Sequence' and it will pop up a
window
> showing the 'dialog ladder' that includes that packet. As you click
on each
> packet in the flow, the main wireshark window will re-position to that
> packet so you can examine it in detail.
>
> Also on the 'SIP Flows' window is a 'Play Streams' button.
Kind of scary
> how easy this is...
>
> --
> Thanks in advance,
> -------------------------------------------------------------------------
> Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867
PST
>             https://www.linkedin.com/in/steve-edwards-4244281
>
> --
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