[OpenSIPS-Users] Slight problem routing 100s and 183s

[Nick Khamis]

Hello Bogdan, thank you so much. The reason for that is the port
forwarding that is from NAT to .5 for SIP and RTP traffic. Do we have
any options to relay the .100 to .5? I tried to set some flags on .5
(i.e., if(status=="183" || status="100")) in the reply_route, but
could not catch the replies coming from .122, even though the sip
trace shows the traffic coming in.

Would opensips just ignore sip traffic with callids it is not aware
of? Finally, could we not force the dialog matching for traffic coming
in from the outside with new callids?

Kind Regards,

Nick.

On 5/23/13, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
> Hi Nick,
>
> if INVITE goes .11 -> .5 -> .10 -> .20 -> .122, why the 100 reply from
> .122 goes to .5 ??? replies have to be relaid back exactly on the same
> path as the request.
>
> So, the 100 reply from .122 must go to .20, not to .5 !!!
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 05/22/2013 08:57 PM, Nick Khamis wrote:
>> Hello Bogdan,
>>
>> Thank you so much for your response, and your time! The log is for the
>> same call, only, the callid is getting changed by asterisk. What is
>> happening is:
>>
>> 192.168.2.11 (UAC) -> 192.168.2.5 (OpenSIPSIn) INVITE
>> Call-ID: 4737d441-5fb15ea7-7142c0d8 at 192.168.2.11
>> <mailto:4737d441-5fb15ea7-7142c0d8 at 192.168.2.11>.
>>
>> 192.168.2.5 (OpenSIPSIn) -> 192.168.2.10 (Asterisk) INVITE
>> Call-ID: 4737d441-5fb15ea7-7142c0d8 at 192.168.2.11
>> <mailto:4737d441-5fb15ea7-7142c0d8 at 192.168.2.11>.
>>
>> 192.168.2.10 (Asterisk) -> 192.168.2.20 (OpenSIPSOut) INVITE
>> Call-ID: 1fbe6fb90553da7c52d72b60076030f5 at 192.168.2.10:5060
>> <http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060>.
>>
>> 192.168.2.20 (OpenSIPSOut) -> 94.101.2.122 (Service Provider) INVITE
>> Call-ID: 1fbe6fb90553da7c52d72b60076030f5 at 192.168.2.10:5060
>> <http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060>.
>>
>> 94.101.2.122:5060 <http://94.101.2.122:5060> (ServicProvider) ->
>> 192.168.2.5:5060 <http://192.168.2.5:5060> (OpenSIPSIn) Giving a Try
>> Call-ID: 1fbe6fb90553da7c52d72b60076030f5 at 192.168.2.10:5060
>> <http://1fbe6fb90553da7c52d72b60076030f5@192.168.2.10:5060>.
>>
>> I am assuming because the callid coming into OpenSIPSIn from the
>> service provider has been changed by asterisk, and OpensipsIn is not
>> aware traffic with that callid, the 183 and 200s are being ignored?
>>
>> I experienced something similar with BYEs and 404, due to changed
>> callid where Vlad solved the problem by explicitly forcing dialog
>> matching using match_dialog. I am not sure if that is possible here too?
>>
>> http://lists.opensips.org/pipermail/users/2013-April/025322.html
>>
>> I also thought about trying to relay the 183 and 200s coming in from
>> the service provider to asterisk. The reason for this is because
>> asterisk has the two callid mapped, and can relay the traffic with the
>> "original" callid back to the proxy.
>>
>> However, to limit the traffic going back and forth, if I can use the
>> "match_dialog" approach again it would be perfect!!
>>
>> This is the last piece of the elephant!!! I hope I can put it together :)
>>
>> Kind Regards,
>>
>> Nick.
>