[Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)
Ing. Angel Gomez Garcia
angom at telnor.net
Sat Aug 23 02:02:07 MST 2003
mmmmmhhh,
Tan Aks wrote:
>For the FXS unit:
> 1) it doesn't recognise voicemail waiting messages, so your analog
>phones won't receive a stuttered dial tone.
>
Right
> 2) it doesn't seem to recognise the transfer (#) button since it seems
>to use different payload numbers (rtp codec 100
>and 96). We will be submitting an email shortly to the bug tracker database.
>
Nop, i do call transfer using #, its working fine.... at least it is for me.
>
>As long as you have the coefficients file defined for your region then call
>handling should be fine. Our gateways are configured for uk use.
>
I have it configured for US and Mexico
>
>Thanks
>Tan
>telappliant.com
>
>
>
>----- Original Message -----
>From: "Ing. Angel Gomez Garcia" <angom at telnor.net>
>To: <asterisk-users at lists.digium.com>
>Sent: Thursday, August 21, 2003 6:39 AM
>Subject: Re: [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway
>(SIP)
>
>
>Hello Ernest.
>I'm setting up two * boxes using mp108 ( FXO and FXS ), and it is
>working fine. The only issue with the FXO box is that it does not
>support remote disconnect supervision so you have to make sure that the
>reorder tone ( wich is used for disconnect ) is adequate for your country.
>
>I have this in sip.conf:
>------------------------------
>
>[mp108out]
>type=friend
>host=x.x.x.x ; <-- Fixed ip assigned to the mp108
>dtmfmode=inband
>
>[mp108in] ; <-- mp108 configured to register with user
>mp108in
>type=friend
>host=dynamic
>dtmfmode=inband
>context=inbound
>
>------------------------------------
>
>And this in extensions.conf
>------------------------------------
>[turnklongdistance]
>exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@mp108out)
>exten => _91NXXNXXXXXX,2,Congestion()
>exten => _91NXXNXXXXXX,102,Congestion()
>
>[longdistance]
>ignorepat => 9
>include => trunklongdistance
>...
>...
>
>[inbound]
>exten => s,1,Wait(2)
>exten => s,2,Answer()
>..... ; Basically your ivr main menu
>
>exten => 100,1,Goto(s|1)
>
>exten => 200,1,Dial(ZAP/2-1)
>.... ; Handling of exceptions
>
>... ; More extensions
>
>-----------------------------------------------
>
>on inbound calls you have to configure the mp108 to forward incoming
>calls to extension 100.
>on outbound calls you have to configure the mp108 to One step dialing.
>
>choose the order of your codecs and i think thats it.
>
>Good luck.
>
>
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