[Asterisk-Users] Grandstream and codec G.711

Mireia Munoz de jesus Mireia.Munoz-de-jesus at insa-lyon.fr
Fri Apr 2 06:32:41 MST 2004


Hi,

My gateway accepts G.711, but not my Grandstream 100 series SIP phone, but I
thought that thanks to CapabilitySet process they will agree to talk with any
other codec, so what is so important G.711 codec?

Best Regards,

Mireia

PS: I am tryaing to get the lattest firmware version, but i don`t know exactly
how to do it. Someone can help?


Quoting Michael Manousos <manousos at inaccessnetworks.com>:

> 
> Hi,
> 
> Mireia Munoz de jesus wrote:
> > Hi!
> > 
> > I have a little big problem here. I have an gateway(asterisk,working as a
> H.323
> > - SIP gateway) conected to a gatekeeper (two different servers), and also
> a
> > gateway (cisco - PSTN) conected to the same gatekeeper. When I make a call
> from
> > the gateway(cisco) to a sip phone, the phone rings, but when I pick up the
> > phone it sounds like it is busy on both sides. 
> > 
> > When I call from SIP to a PSTN phone, I have the next error on written on
> CLI:
> >  
> > -- Executing Dial("SIP/03316410-a668", "OH323/483317839|20|t|T") in new
> stack
> 
> First, this should be:
> Dial(OH323/483317839|20|tT) not
> Dial(OH323/483317839|20|t|T)
> 
> > [2]WrapperAPI::h323_make_call: Making call.
> > [2]WrapMutex::Wait: Requesting mutex callMutex [wrapendpoint.cxx, 269,
> MakeCall]
> > [2]WrapMutex::Wait: Got mutex callMutex [wrapendpoint.cxx, 269, MakeCall]
> > [2]WrapH323EndPoint::MakeCall: Making call to 483317839
> > [3]WrapH323Connection::WrapH323Connection: Outgoing capability
> > G.711-uLaw-64k{hw}
> > [3]WrapH323Connection::WrapH323Connection: Caller ID name on outgoing call
> > Felippe
> > [3]WrapH323Connection::WrapH323Connection: LocalPartyName Felippe
> > [3]WrapH323Connection::WrapH323Connection: DestExtraCallInfo
> > [3]WrapH323Connection::WrapH323Connection: Caller ID on outgoing call
> 03316410
> > [3]WrapH323EndPoint::MakeCall: Call token is ip$localhost/5074
> > [3]WrapH323EndPoint::MakeCall: Call reference is 5074
> > [2]WrapMutex::Signal: Released mutex callMutex [wrapendpoint.cxx, 320,
> MakeCall]
> > [2]WrapH323Connection::OnSendSignalSetup: Sending SETUP message...
> > [3]WrapH323Connection::OnSendSignalSetup: Setting display name Felippe
> > [3]WrapH323Connection::OnSendSignalSetup: Setting calling party number
> 03316410
> > [2]WrapH323Connection::OnReceivedReleaseComplete: Received RELEASE
> COMPLETE
> > message [ip$localhost/5074]
> > [2]WrapH323EndPoint::ClearCall: Request to clear call [ip$localhost/5074]
> > [2]WrapH323EndPoint::ClearCall: Request to clear call [ip$localhost/5074]
> > [2]WrapH323EndPoint::ClearCall: Request to clear call [ip$localhost/5074]
> > [2]WrapH323EndPoint::OnConnectionCleared: Connection [ip$localhost/5074]
> closed.
> 
> It looks like a codec conflict. Are you sure that G.711 is allowed
> by the remote Cisco?
> 
> > [2]WrapMutex::Wait: Requesting mutex channelMutex [wrapendpoint.cxx, 967,
> > OnConnectionCleared]
> > [2]WrapMutex::Wait: Got mutex channelMutex [wrapendpoint.cxx, 967,
> > OnConnectionCleared]
> > [2]WrapH323EndPoint::OnConnectionCleared: Call with "150.xxx.xxx.xxx"
> completed
> > [2]WrapMutex::Signal: Released mutex channelMutex [wrapendpoint.cxx, 1047,
> > OnConnectionCleared]
> > Mar 31 10:39:58 ERROR[311316]: chan_oh323.c:1135 oh323_call: OH323/L0:
> Could not
> > call 483317839.
> >     -- Couldn't call 483317839
> > [2]WrapperAPI::h323_clear_call: Clearing call.
> > [3]ClearCallThread::ClearCallThread: Unblock pipe - 46, 47
> >     -- Hungup 'OH323/L0'
> >   == Everyone is busy at this time 
> > 
> > ANyone has any idea of what is going on?
> > 
> > Best Regards,
> > 
> > Mireia
> 
> Michael.
> 
> 
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