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: [email protected]. 192.168.2.5 (OpenSIPSIn) -> 192.168.2.10 (Asterisk) INVITE Call-ID: [email protected]. 192.168.2.10 (Asterisk) -> 192.168.2.20 (OpenSIPSOut) INVITE Call-ID: [email protected]:5060. 192.168.2.20 (OpenSIPSOut) -> 94.101.2.122 (Service Provider) INVITE Call-ID: [email protected]:5060. 94.101.2.122:5060 (ServicProvider) -> 192.168.2.5:5060 (OpenSIPSIn) Giving a Try Call-ID: [email protected]: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.
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