答复: sip transfer question

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Hi!
I have this problem too.
Can you fix it already?

CharlesFan
BestRegrads

-----????-----
???: pjsip-bounces at lists.pjsip.org [mailto:pjsip-bounces at lists.pjsip.org] ?? carlos calvo
????: 2010?5?19? 17:48
???: pjsip at lists.pjsip.org
??: [pjsip] sip transfer question

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.

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