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I have some troublesome numbers that I would like to capture the SIP dialogue when I am calling them. When I am about to dial the number, is there any way to turn on SIP debugging in the dial plan before I make the call? (and turn it off after the call is completed?) -------------- next part -------------- An HTML attachment was scrubbed... URL:

2008/11/4 Igor Zamocky <asterix at ponozky.sk> > > Did You tried http://www.voip-info.org/wiki/view/Asterisk+sip+limitonpeers > ? > I didn't. Now I did and it's working the way I wanted. Meanwhile, I had found a (complex) workaround using GROUP, GROUP_COUNT and SIPPEER but limitonpeers is mu...

Hello, I've got strange problems trying to run asterisk with MGCP ip phones (Thomson ST2030). Situation: "user A" <----- pstn -------> ASTERISK <----- mgcp ------> "user B" "User A", connected behind a PSTN, tries to call "User B". After dialing "User B"'s number, call comes to ASTERISK, ASTERISK contacts

Hello, I've problems with following - ----- --- --- PSTN | --- isdn --- | A | ----- iax2 ------ | B | ----- --- --- On [B], there is unconditional call forwarding set back via [A] (dialparties.agi is used) to PSTN. So, call from PSTN is routed via [A] to [B] and than back again into PSTN.