[OpenSIPS-Users] Identify an in progress call

[Bogdan-Andrei Iancu]

Hi guys,

The normal approach will be to do t_newtran() asap in your script -> 
this will take care and absorb retransmissions, but has the downside of 
creating the transaction at that moment, so whatever changes you do 
later (over the message) will not be recorded into transaction (if you 
need them in failure route, for serial forking).

Regards,
Bogdan

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 08/16/2012 11:52 AM, Muhammad Shahzad wrote:
> I had a similar problem a while ago and i deal with it in my own way, 
> which was quite simple actually.
>
> 1. First INVITE comes in, i check it does not has to_tag, i put its 
> call-id in memcache and do normal processing.
> 2. For subsequent INVITE without to_tag, i check in memcache if 
> call-id already exists, if yes then send stateless reply 100-Trying 
> and exit. Otherwise treat it as first INVITE.
> 3. Memcache auto expires each record after 60 seconds, so we have auto 
> clean up regardless if call was successful or failed.
>
> If there is any better way, i would be happy to learn it.
>
> Thank you.
>
>
> On Thu, Aug 16, 2012 at 7:25 AM, Andrew Mason 
> <andrew.mason at voice.net.au <mailto:andrew.mason at voice.net.au>> wrote:
>
>     Hi All,
>     We recently encountered an issue in our datacenter where an
>     upstream switch
>     between the CPE and the OpenSIPS server was dropping packets.
>     Obviously for
>     VoIP this is not ideal but it did expose an issue in our
>     configuration.
>
>     If the packet dropped was a '183 Session Progress' or a 'Ringing'
>     then the CPE
>     would send the same Invite (exact same packet) to which our server
>     issues a
>     407 Proxy Auth Required. At which stage the CPE seems to hang up.
>
>     My question: How can I determine if a call is already in progress
>     for a
>     particular Invite ?
>
>     Apologies if my terminology is not correct, please let me know if
>     anything
>     does not make sense. I have inherited this system and am still
>     very much
>     learning about SIP and OpenSIPS.
>
>     Thanks in advance
>     Andrew
>
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>
>
>
> -- 
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com <mailto:shari_786pk at hotmail.com>
> Email: shaheryarkh at googlemail.com <mailto:shaheryarkh at googlemail.com>
>
>
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