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

Thu Jan 10 11:06:20 CST 2008

I appreciate your answer and really think that altering button  
functions is the way to go on Polycom phones. But I also mentioned  
that when using ZAP FXS channels and transferring using FLASH the  
problem also appears.

Is there a way to map the FLASH on FXS channels to the sequence  
configured on features.conf?

Att
Vinícius Fontes
Desenvolvimento
Canall Tecnologia em Comunicações

Em 10/01/2008, às 12:50, Michael Cargile escreveu:

> 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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