[asterisk-dev] MixMonitor doesn't work right with SIP and Zap/Flash transfers

[Michael Cargile]

I wont speculate on whether this is a bug or not but I do know that with
Polycom phones you can change the behavior some of buttons transfer
being one of them. IE change it so it sends a # to asterisk to initiate
a transfer.

You have to use polycoms provisioning system and deal with their
monstrous XML based config file that many people find daunting, but if
you read through their admin guide and search around online you will
find a way to do it.

Michael Cargile
Software Developer
Explido Software USA Inc.
www.explido.us

On Tue, 2008-01-08 at 16:45 -0200, Vinicius Fontes wrote:
> Hey guys, I don't know if this is the right place to ask this. I was  
> thinking about reporting a bug, but maybe it's better to sort out if  
> this is really a bug or just me being lame.
> 
> I want to record *every* call in my Asterisk box, so I use the  
> MixMonitor() application like this is my extensions.conf:
> 
> exten => _0X.,1,Answer()
> exten => _0X.,n,MixMonitor(${CALLERID(num)}-${STRFTIME($ 
> {EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
> exten => _0X.,n,Dial(IAX2/pabx-canall/${EXTEN},60,tT)
> 
> exten => _2XX,1,Answer() exten => _2XX,n,MixMonitor(${CALLERID(num)}-$ 
> {STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)  
> exten => _2XX,n,Dial(SIP/${EXTEN},60,tT)
> 
> The scenario is as following:
> 
> 1) 201 asks operator for an external call, hangs up. The audio file is  
> stored correctly. From the CLI:
> 
> [Jan 8 16:20:19] -- Executing [200 at default:1] Answer("SIP/ 
> 201-081d8740", "") in new stack
> [Jan 8 16:20:19] -- Executing [200 at default:2] MixMonitor("SIP/ 
> 201-081d8740", "201-2008-01-08-16-20-19-200.wav") in new stack
> [Jan 8 16:20:19] -- Executing [200 at default:3] Dial("SIP/201-081d8740",  
> "SIP/200|60|tT") in new stack
> [Jan 8 16:20:19] == Begin MixMonitor Recording SIP/201-081d8740
> [Jan 8 16:20:19] -- Called 200
> [Jan 8 16:20:19] -- SIP/200-081fac90 is ringing
> [Jan 8 16:20:23] -- SIP/200-081fac90 answered SIP/201-081d8740
> [Jan 8 16:20:27] == Spawn extension (default, 200, 3) exited non-zero  
> on 'SIP/201-081d8740'
> [Jan 8 16:20:27] == End MixMonitor Recording SIP/201-081d8740
> 
> 
> 
> 
> 2) 200 dials to the PSTN. So far so good.
> 
> [Jan 8 16:20:35] -- Executing [021047020 at default:1] Answer("SIP/ 
> 200-081d8740", "") in new stack
> [Jan 8 16:20:35] -- Executing [021047020 at default:2] MixMonitor("SIP/ 
> 200-081d8740", "200-2008-01-08-16-20-35-021047020.wav") in new stack
> [Jan 8 16:20:35] -- Executing [021047020 at default:3] Dial("SIP/ 
> 200-081d8740", "IAX2/pabx-canall/021047020|60|tT") in new stack
> [Jan 8 16:20:35] == Begin MixMonitor Recording SIP/200-081d8740
> [Jan 8 16:20:35] -- Called pabx-canall/021047020
> [Jan 8 16:20:35] -- Call accepted by 200.248.136.140 (format alaw)
> [Jan 8 16:20:35] -- Format for call is alaw [Jan 8 16:20:35] -- IAX2/ 
> pabx-canall-16384 answered SIP/200-081d8740
> 
> 
> 
> 
> 3) Extension 200 is a Polycom SoundPoint 301 IP phone. It presses the  
> Transfer button, putting 021047020 in hold and dialing to 201 who  
> answers the call:
> 
> [Jan 8 16:20:45] -- Started music on hold, class 'default', on IAX2/ 
> pabx-canall-16384
> [Jan 8 16:20:51] -- Executing [201 at default:1] Answer("SIP/ 
> 200-081fac90", "") in new stack
> [Jan 8 16:20:51] -- Executing [201 at default:2] MixMonitor("SIP/ 
> 200-081fac90", "200-2008-01-08-16-20-51-201.wav") in new stack
> [Jan 8 16:20:51] -- Executing [201 at default:3] Dial("SIP/200-081fac90",  
> "SIP/201|60|tT") in new stack
> [Jan 8 16:20:51] -- Called 201
> [Jan 8 16:20:51] == Begin MixMonitor Recording SIP/200-081fac90
> [Jan 8 16:20:51] -- SIP/201-081edf80 is ringing
> [Jan 8 16:20:54] -- SIP/201-081edf80 answered SIP/200-081fac90
> 
> 
> 
> 
> 4) The operator says "here's your call" to 201 and presses Transfer on  
> the phone once more. The call is transferred correctly, but:
> [Jan 8 16:20:57] -- Stopped music on hold on IAX2/pabx-canall-16384
> [Jan 8 16:20:57] == Spawn extension (default, 021047020, 3) exited non- 
> zero on 'SIP/200-081d8740'
> [Jan 8 16:20:57] == End MixMonitor Recording SIP/200-081d8740
> [Jan 8 16:20:57] == End MixMonitor Recording SIP/200-081fac90
> 
> 
> Notice that all the MixMonitor processes stopped!
> 
> 
> 
> 5) 201 finally hangs up the phone:
> 
> [Jan 8 16:21:45] == Spawn extension (default, 201, 3) exited non-zero  
> on 'IAX2/pabx-canall-16384'
> [Jan 8 16:21:45] -- Hungup 'IAX2/pabx-canall-16384'
> 
> 
> 
> So, all the audio regarding the important part -- the call to the PSTN  
> itself -- is simply lost.
> 
> I noticed that if I use Asterisk's built-in transfer features (atxfer,  
> blindxfer) everything works fine. Too bad the users are so used to  
> that Transfer button. I tried it using FXS channels and the FLASH  
> button on the phone, same results.
> 
> Is there any workaround for this? I'm running these from a separate  
> box so any procediment you guys could suggest will be tried as it is  
> not in production. I'm also willing to give you any information needed.
> 
> Thanks in advance.
> 
> 
> Att
> Vinícius Fontes
> Desenvolvimento
> Canall Tecnologia em Comunicações
> 
> 
> 
> 
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