hello, i'm currently using asterisk (cvs-head) as PSTN gateway. the routing logic is mostly done in OpenSER. the problem is that i'm not able to transfer calls between the PSTN and another SIP peer (when the PSTN<>SIP connection goes over asterisk but the SIP<>SIP connection does not).
there are 2 possibilities for asterisk to be part of the transfer: 1) asterisk receives the REFER msg. in this case asterisk logs the following error msg: "Supervised transfer requested, but unable to find callid '%s'. Both legs must reside on Asterisk box to transfer at this time." 2) the REFER goes to the other peer - asterisk just receives the INVITE: this basically works but it seems that asterisk ignores the 'replaces' header. therefore a new call is initiated instead of a transfer of the current call. i've checked the STABLE src and it seems to be affected by this problem too (i haven't tested it - just skimmed through the src). the only solution for transfers to work seems to be to route all calls over asterisk - which is a solution i'd like to avoid because of scalability and compability (it breaks ICE) issues. any other suggestion how to get transfers to work without routing _all_ the calls through asterisk? /gst
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