atis at iq-labs.net
Guest
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 Posted: Wed Jan 09, 2008 10:11 am Post subject: [asterisk-users] [asterisk-dev] MixMonitor doesn't work righ |
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Vinicius Fontes wrote:
| Quote: | 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.
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Make sure you have "canreinvite=no" for peers, that will ensure that RTP
is always passed trough asterisk. Now your Polycom might send audio
directly to other phone.
Btw, asterisk-dev is for development discussions, but this is
configuration problem. If unsure, you should write to asterisk-users
first (cross-posted there)
Regards,
Atis |
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