Home
Software
VoIP Server
WebPhone
Softphones
Customized Softphone
Windows softphone
Android Softphone
iPhone Softphone
Symbian Softphone
Others
SIP SDK
Java SIP SDK
Windows SIP SDK
Android SIP SDK
iOS SIP
Web SDK
All SIP SDK
VoIP Tunnel
WebRTC to SIP
VoIP Push Gateway
VoIP Tester
SIP Load Balancer
More
Solutions
All in one
Wholesale platform
Click to call
VoIP Billing
VoIP Push
WebRTC
Codec transcoding
VoIP call recording
NAT solution
Presence and IM
IVR
IP Centrex
Predictive dialer
SIP client engine
VoIP Encryption
Large scale VoIP
Services
VoIP Hosting
VoIP Services
JavaScript Phone
VoIP Development
Support
Contact
Documentations
Download
Company
About us
Contact
News
Softphone, webphone and VoIP server Forum
All Forums
1
2
3
4
5
Home
Discussions
Mizu Webphone
Recording vocal messages with Mizu
Previous
Next
9/5/2012 7:11 AM
Domotica Labs
Joined: 9/5/2012
Posts: 1
Recording vocal messages with Mizu
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 ?
10/1/2012 7:09 AM
SuperUser Account
Joined: 5/9/2008
Posts: 481
Re: Recording vocal messages with Mizu
Hello
Please contact info@mizu-voip.com if you still have this issue.
10/10/2012 4:05 AM
firionicable
Joined: 10/10/2012
Posts: 20
Re: Recording vocal messages with Mizu
You should contact support via email for this.
Works like a charm..
Page 1 of 1
Previous
Next
Home
Discussions
Mizu Webphone
Recording vocal messages with Mizu
You need to
register
/
login
to be able to post in the forum
Forum home
Resources
VoIP Forum home
Webphone
Java SIP client
Java SIP client
Softphone for Android
VoIP Hosting
All-In-One VoIP solution
Softswitch
Windows softphone
Customized Softphone
Documentation
Download
Wiki
Blog
Email to support
send email
Home
|
Software
|
Solutions
|
Services
|
Support
|
Company
Privacy Statement
|
Terms Of Use
Copyright (©) 2023 Mizutech S.R.L.