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I am using two SIPml5 demo + asterisk to make a call each other.

I can hear the sound from one end but can't from the other end.

I succeed once and suddenly lost one side after some changes i don't remember at all.

I think I am using the same configuration/setting but can't figure out this issue.

The versions I am using is.. Asterisk: 11.10.0 SIPml5: latest chrome: 33.0.1750.152

I am pretty sure I was using the chrome 33.0.1750.152

rtp.conf

[general]
rtpstart=8000
rtpend=8100
icesupport=yes
stunaddr=stun.l.google.com:19302

sip.conf

[general]
udpbindaddr=0.0.0.0:5060
transport=ws

[4000]
type=friend
host=dynamic
secret=1234
context=sipml5
encryption=yes
avpf=yes
icesupport=yes
allow=ulaw
permit=192.168.1.0/255.255.255.0

[4001]
type=friend
host=dynamic
secret=1234
context=sipml5
encryption=yes
avpf=yes
icesupport=yes
allow=ulaw
permit=192.168.1.0/255.255.255.0

extensions.conf

[sipml5]
exten=>1000,1,Answer()
same=>n,Playback(demo-congrats)
same=>n,Hangup()

exten=>4000,1,Dial(SIP/4000)
exten=>4001,1,Dial(SIP/4001)

two users are 4000 and 4001 and when I dial 1000, I can hear demo-congrats playback for both.

Please do not say "it's highly experimental".. I did it before which means it's possible.

Shevliaskovic
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haeminish
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  • why nobody answered that? I have the exact same problem – Alireza Fallah Aug 10 '14 at 09:59
  • I also don`t hear any sound from Playback App when using Asterisk 11.12 and Chrome (36 and 38beta). But in Asterisk 13-beta1 everything works and I can hear sound from asterisk with same configuration. I do not use example client, but write my own using sipml5 and [api guide](http://sipml5.org/docgen/index.html?svn=224) – gkocjan Sep 05 '14 at 09:05
  • Can you check your RTP log by executing rtp set debug on from the CLI? Make sure you have your rtp packets sent to the correct ip address (your sipml5 running computer's ip) and you receive packets from the same too. (You should see a RTP from and RTP to consecutivey) – Sachin Murali G Nov 18 '14 at 07:15

1 Answers1

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In some cases there's an audio issue if you don't Answer the call. You could try this in your dialplan

exten=>4000,1,Answer()
same => n,Dial(SIP/4000)
exten=>4001,1,Answer()
same => n,Dial(SIP/4001)

I would also recommend to use patterns instead of duplicating code:

exten => _400[01],1,Answer()
same => n,Dial(SIP/${EXTEN})
tpalanques
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