I just verified it here as well. Running Asterisk 1.2.11 and two polycom
phones running 1.6.7 firmware with canreinvite=yes. Putting the call on
hold and then off does bring the audio back. From what I can tell by
looking at the lights on my switch, something is still sending the RTP
traffic to the phone that made the transfer even after it has hung up.
Once the transferee phone places the call on hold and then picks it back
up that traffic stops. This was tested with a call IAX -> SIP1 where
SIP1 does an attended transfer to SIP2.
Doing a call from SIP1->SIP2 and then transferring SIP2 to SIP3 works
just fine.
-Dave
Kevin Smith wrote:
Hi Avi,
I had a similar problem. Have extension 405 put the call on hold (after
the transfer) and then off hold. I am willing to bet it will bring back
the audio stream. I posted something similar a few weeks ago and if
anyone thought it was a bug, to let me know what information I needed to
send in to report it, but no one replied.
Anyway, I noticed it happening on the latest release of asterisk. I
rolled back my installation so I am on asterisk 1.2.9.1, lib 1.2.3, and
zaptel 1.2.6 and that corrected the problem for me.
Kevin
Avi Miller wrote:
Hey guys,
I've been trying to change my Asterisk setups to use canreinvite=yes.
I'm having a small problem with my Polycom IP501 phones and
transferring calls.
If a call comes in via my ISDN BRI lines (using chan-capi), I can
successfully transfer the call using the Polycom Blind Transfer option
(Transfer -> Blind -> EXT -> Send).
However, if I try to use the attended transfer method, the call is
never connected to the new user. When I hit transfer, the caller gets
MOH and I dial the destination ext. Once the person answers, I hit
"Transfer"
Now .. the MOH stops for the caller, but both phones are dead. The
call is never reconnected successfully. On the console, I see this:
-- Called 405
-- SIP/405-0849cba0 is ringing
-- SIP/405-0849cba0 answered SIP/401-084a0ba8
-- Attempting native bridge of SIP/401-084a0ba8 and SIP/405-0849cba0
-- Stopped music on hold on CAPI/V4BRI-2/92355400-25
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/401-084a0ba8<ZOMBIE>' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/401-084a0ba8<ZOMBIE>'
-- Incoming call: Got SIP response 500 "Internal Server Error"
back from 192.168.1.128
== Spawn extension (macro-dial, s, 10) exited non-zero on
'CAPI/V4BRI-2/92355400-25' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'CAPI/V4BRI-2/92355400-25' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'CAPI/V4BRI-2/92355400-25'
405 is the extension I'm trying to transfer the call to.
Any advice? I've been searching the list archives and the wiki, but
can't find anything specific.
Ta,
Avi
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