Re: [asterisk-users] no audio on end-point when call is connected/bridged via PBX

2010-12-07 Thread Robles Román , José Miguel

 I am running Asterisk 1.8 on a cloud server.  I have had the
 same configuration running on a physical machine with a
 similar configuration.
 Thoughts?  I know I posted this yesterday but was hoping for
 some more creative comments!

If signalling works and audio don't, it probably has to do with phones behind 
NAT. It seems necessary to review the configuration of local routers.

Regards,
José Miguel

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[asterisk-users] no audio on end-point when call is connected/bridged via PBX

2010-12-06 Thread Thomas Perron
I am trying to dial through my asterisk machine from phone A to phone B.
My DID is registered properly with the SIP provider.  When I dial from
A to B it looks fine so far.
A rings B and B can pick up and the call is bridged.  However, I don't
hear any audio so therefor it is not working.
I am running Asterisk 1.8 on a cloud server.  I have had the same
configuration running on a physical machine with a similar
configuration.
Thoughts?  I know I posted this yesterday but was hoping for some more
creative comments!

Zip*CLI sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
sip.callwithus.com:5060 N    105
Registered   Tue, 07 Dec
2010 02:36:43
1 SIP registrations.

my sip.conf
[general]
context=default
allowoverlap=no
;bindport=5060
port=5060
bindaddr=0.0.0.0
canreinvite=no ;if your asterisk box is behind a NAT ro

;register = :3...@carrier.callwithus.com
register = :3...@sip.callwithus.com

[callwithus]
type=friend
host=sip.callwithus.com
username=
secret=31
qualify=no
insecure=invite


my extensions.conf
[general]

[globals]
CONSOLE=Console/dsp ; Console interface for demo
OUTBOUNDTRUNK=SIP/callwithus

[default]
exten = s,1,Answer()
exten = s,n,Dial(SIP/callwithus/122)
exten = s,n,Wait(2)
exten = s,n,Hangup()

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