Displaying 3 results from an estimated 3 matches for "ponozki".
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ponozky
2008 Nov 04
0
Is SIPPEER curcalls working for you ? [SOLVED]
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 much more concise.
Thanks a lot.
>
>
> Hi,
>
>
> In this
2008 Jan 21
0
MGCP Thomson, "early" transmit problem
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
2006 Feb 13
1
asterisk still tries native bridging
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.