[Serusers] CANCEL request

[Klaus Darilion]

Greger V. Teigre 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.

AFAIK there is an ACK to every 4xx response.

klaus

> 
>> 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-)
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