[Serusers] CANCEL request

[Iqbal]

coll so its fine,

As for the comparision, I think good idea also, just did one for call
transfers between phones, man I have pages of it :-)

I have sip_scenario installed to show what is happeing, but even that can
be confusing, ngrep dump and compare for mosr will be easier.

Replacing the times, and domains should not a be a problem.

1. save ngrep to output file
2. run rep.pl I could write a few lines for this, use sample.com as the
domain, and try to use a regex to replace the existing domain (havent
tried this so I might be jumping in and getting burnt)
3. Then need a standard dump from basic onsip setup, if you have one with
a sip to sip call, and sip to pstn send me a copy, and I'll run my
setup and see if the proggie does a good compare.

iqbal

On 6/29/2005, "Greger V. Teigre" <greger at teigre.com> wrote:

>See inline.
>
>Iqbal wrote:
>> Hi
>>
>> I have pstn -->ser -->UA
>>
>> I also have asterisk hanging off ser for voicemail
>>
>> Now this is all working fine, voicemail and all triggers great on no
>> answer, BUT to be sure I decided to look atthe sip dialogue, just to
>> see if all was fine, and so that I could start to clean up my config
>> file.
>> When a user calls from pstn, then hit the switch, and drop into ser,
>> which then maps the pstn number to a local alias.
>> Ip phone rings,
>>
>> INVITE (pstn) to (ser)
>> 100 trying (ser) to (pstn)
>> INVITE (ser) to (ua)
>> 100 trying (ua) to (ser)
>> 180 ringing (ua) to (ser)
>> 180 ringing (ser) to (pstn)
>>
>> so far so good, now if there is no answer, and I forward to asterisk
>> should there be a cancel to the original INVITE, cause this is what I
>> am getting:
>>
>> CANCEL (ser) to (ua)
>> 200 OK (ua) to (ser)
>> 487 request cancelled (ua) to (ser)
>>
>> then i get
>> ACK (ser) to (ua) ------where this ACK comes fro I am not sure
>I believe the 487 does not need an ACK, but that the ACK is for the OK from
>ua.
>
>> 200OK (ser) to (pstn) but useragent is no asterisk, hence this makes
>> sense
>This OK should be from Asterisk, right?! This OK will contain the SDP from
>Asterisk
>
>>ACK (pstn) to (ser)
>This should go to Asterisk
>>
>> so what I am not clear on is should the CANCEL be there, or not, it
>> seems to make sense that it is, just want to confirm.
>
>Yes, when the timer goes off, SER will cancel the INVITE.  This is correct.
>
>> Also since alot of people have the same setup, would it be a good idea
>> alongside onsip.org and its startup config, if we could post/have a
>> sip trace of common call scenarios, I know some of these are in the
>> rfc etc, but someone they dont seem user friendly...
>
>Yes, good idea.  We have been toying with the idea of standardized call
>scenarios using sipp that we can use for onsip.org testing to verify that
>everything works after changes have been done.  We have focused on getting
>new issues out...
>    So, if somebody would like to make some standard sipp scenarios that can
>be played using a script, we can certainly generate SIP traces for each
>config file as a reference.  We need to remove the time added by ngrep and
>standardize on a username/domain, as well as do a replacelement on IP
>addresses, so that one can use diff to view any differences. :-)
>
>g-)
>
>
>