Hi! following problem, i guess this is more a SIP protocol issue than a pjsip, but still, maybe someone can give me a good hint how to solve the problem I've done some testing with pjsip-ua configuring two accounts, see configuration below. $ pjsip-ua --config-file test001.cfg $ cat test001.cfg --registrar sip:192.24.6.23 --id sip:101 at 192.24.6.23 --contact sip:101 at 173.24.1.105:5060 --contact-params user=phone --contact-uri-params user=phone --username 101 --password 1111 --realm node000000 --null-audio --reg-timeout 3600 ?--next-account --registrar sip:192.24.6.23 --id sip:102 at 192.24.6.23 --contact sip:102 at 173.24.1.105:5060 --contact-params user=phone --contact-uri-params user=phone --username 102 --password 2222 --realm node000000 --null-audio --reg-timeout 3600 I set up two different calls with the each of the configured accounts (101 and 102) to two different target extensions (to 201 and 202 respectively). At this point, i would like to, somehow, join/transfer those calls, so 201 and 202 can talk each other directly and 101 and 102 are out. If i use the transfer command 'X' (with Replaces) i get back (i am using a omniPCX for testing) error code 488. (Not acceptable here) Not sure wether transfer (REFER) is the proper solution to solve this problem. Does SIP provide any means to solve this? I though about a 'two step' transfer: ?- 101 talks to 201 and 102 talks to 202 ?- 101 puts on hold 201 ?- 102 puts on hold 202 ?- 101 calls to 102 ?- 101 transfers 201 to 102 ?and now a normal attended transfer: ?- 102 transfer 201 to 202 i haven't really been able to test this with pjsip-ua but i was wondering whether there is a more straightforward way to cope with this problem. thanks Carlos.
