On Wed, 2006-02-22 at 11:51 +0100, Daniele orlandi wrote: > > But if gsm hangups first, asterisk doesn't send a signal to > > hangup, therefore sip client still thinks that call is active. > > I see the extension exiting correctly, I don't know why your SIP phone > doesn't > see this hangup: > > == Spawn extension (visdn0_in, 7510078, 1) exited non-zero on > 'VISDN/visdn0/1.I'
This line is printed AFTER SIP phone hangups. So if I hangup on ISDN line first, nothing happens. Asterisk still thinks that call is in progress and I can hear a congestion/busy signal that comes from telco. > > But, I still don't know why call is handled with > > exten => s,1,Dial(IAX2/102) when it comes from ISDN > > and with exten => _751007[78],1,Dial(IAX2/102) > > when it comes from GSM? > > Because GSM calls are always en-bloc (sending complete) while calls coming > from ISDN may be both (in your case they use overlap receiving). > > Anyway, why are you putting Dial() in the 's' extension? You should have > VISDNOverlapDial() there. > Seems like I learned a new thing. If I understand correctly VISDNOverlapDial() waits for sending digits to complete or timeout and then executes appropriate extension? thank you > Bye, > _______________________________________________ > Visdn-hackers mailing list > [email protected] > https://mailman.uli.it/mailman/listinfo/visdn-hackers _______________________________________________ Visdn-hackers mailing list [email protected] https://mailman.uli.it/mailman/listinfo/visdn-hackers
