[Asterisk-Users] Audiocodes FXO - loop channels
Daniel Cubero Salas, Ing
dcubero at cosinet.net
Wed Feb 4 16:48:48 MST 2004
Hi all..!!
We have a trouble with asterisk 0.4.0 and audiocodes MP-108 FXO.
For someone reason some times one extension dial a number which is a PSTN
call (use audiocodes for outcalling but incalling case audiocodes send this
call to one extensions in asterisk like IVR), when that call was finished
audiocodes don´t release the call and line FXO port but send a request for
that phone number (the outcalling contest is different from incalling
contest) then asterisk take that number and request to audiocodes a
outcalling with that phone number, who use a other FXO port... and then a
loop is showed, the worst is never release that 2 FXO ports... I don´t sure
that audiocodes is whom send first request... but that stuff is random
Above a brief report from CLI using show channels:
Channel (Context Extension Pri ) State Appl. Data
SIP/audiocodes-875b (sip_out 1 ) Up Bridged Call
SIP/audiocodes-78cb
SIP/audiocodes-78cb (sip 2XXXXXX 6 ) Up Dial
SIP/2XXXXX at audiocodes|300|Tr
2 active channel(s)
"XXXXX" is a random phone number
Where is the mistake? Can anyone help us?
Sorry for my broken english :)
Regards
Daniel
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