Kind technical support,

I am trying to use your Mizu Voip client to call an asterisk server, register a simple vocal message and then playback it.
Right now I am using a Mizu client version: 4.0.1 and an asterisk server version: 1.8.8.2

My asterisk's extension configuration files is really simple:
exten => *77,1,Answer
exten => *77,2,Wait(2)
exten => *77,3,Record(/tmp/asterisk-recording:wav,3)
exten => *77,4,Wait(2)
exten => *77,5,Hangup
exten => *99,1,Answer
exten => *99,2,Wait(2)
exten => *99,3,Playback(/tmp/asterisk-recording)
exten => *99,4,Wait(2)
exten => *99,5,Hangup

I simply want to call *77, register a message, stop the registration after 3 seconds of silence and then call *99 to listen it.

If I use other voip clients I don't have any problem.
The correct asterisk log during recording is:
== Using SIP RTP CoS mark 5
    -- Executing [*77@local:1] Answer("SIP/104-0000003a", "") in new stack
    -- Executing [*77@local:2] Wait("SIP/104-0000003a", "2") in new stack
    -- Executing [*77@local:3] Record("SIP/104-0000003a", "/tmp/asterisk-recording:wav,3") in new stack
    -- <SIP/104-0000003a> Playing 'beep.gsm' (language 'en')
    -- Executing [*77@local:4] Wait("SIP/104-0000003a", "2") in new stack
    -- Executing [*77@local:5] Hangup("SIP/104-0000003a", "") in new stack

If I try to use Mizu, I can hear the playback but I am unable to record the message correctly: it seems to record forever without being able to stop the registrarion after 3 seconds of silence.

If I check the asterisk log I see two warning and asterisk looping on the recording action:
 == Using SIP RTP CoS mark 5
    -- Executing [*77@local:1] Answer("SIP/102-00000037", "") in new stack
[Sep  4 16:49:09] WARNING[5078]: res_rtp_asterisk.c:2060 ast_rtp_read: RTP Read too short
[Sep  4 16:49:09] WARNING[5078]: res_rtp_asterisk.c:2060 ast_rtp_read: RTP Read too short
    -- Executing [*77@local:2] Wait("SIP/102-00000037", "2") in new stack
    -- Executing [*77@local:3] Record("SIP/102-00000037", "/tmp/asterisk-recording:wav,3") in new stack
    -- <SIP/102-00000037> Playing 'beep.gsm' (language 'en')

While the log on Mizu ends with:
...
[167-880] [17:23:30:162] EVENT,set state from 13 [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:163] EVENT,switching state from InCall to Speaking [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:165] EVENT,set deleteat 443 to 10920000 [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:171] REC,192.168.0.105:5060 on port 18070
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.56:18070;branch=z9hG4bK-1445p8788876277235122737r;received=192.168.0.56;rport=18070
From: <sip:102@192.168.0.105>;tag=1444g3482740103147970405m
To: <sip:*77@192.168.0.105>;tag=as72899230
Call-ID: 1442e5725131561578430932k13127rmwp
CSeq: 2551 INVITE
Server: Asterisk PBX 1.8.8.2~dfsg-1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:*77@192.168.0.105:5060>
Content-Type: application/sdp
Content-Length: 265


v=0
o=root 27953310 27953310 IN IP4 192.168.0.105
s=Asterisk PBX 1.8.8.2~dfsg-1
c=IN IP4 192.168.0.105
t=0 0
m=audio 19006 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
 [mt: 61032]
[167-880] [17:23:30:176] EVENT,old ep found for message [mt: 61032]
[167-880] [17:23:30:178] EVENT,cseq set1 to 2551 [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:179] EVENT,set state from 3 [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:183] EVENT,switching state from Speaking to Midcall [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:184] EVENT,set deleteat 443 to 10920000 [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:185] EVENT,unknown ok received [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:186] EVENT,build uri rule 13 [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:30:187] SEND:  to 192.168.0.105:5060 from port 18070
ACK sip:*77@192.168.0.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.56:18070;branch=z9hG4bK-1445p8788876277235122737r
From: <sip:102@192.168.0.105>;tag=1444g3482740103147970405m
To: <sip:*77@192.168.0.105>;tag=as72899230
Call-ID: 1442e5725131561578430932k13127rmwp
CSeq: 2551 ACK
Max-Forwards: 35
Contact: <sip:102@192.168.0.56:18070>
User-Agent: MizuWebPhone/4.0.1
FinalUA: MizuWebPhone
Content-Length: 0


 [mt: 61032]
[167-880] [17:23:32:199] EVENT, set rtp rec address to 192.168.0.105:19006 [ep: 1442e5725131561578430932k13127rmwp 61032]
[167-880] [17:23:32:200] EVENT, (rtp)  marker received 1
[167-880] [17:23:32:221] EVENT,prebuffering finished (0) at 2 jcount, 640 bytes

Do you have any advice to help me solve such event ?

You should contact support via email for this.

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Works like a charm..