[asterisk-users] Polycoms, Attended Transfer and Canreinvite = yes

[Dave Fullerton]

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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