#asterisk - Tue 20 Mar 2007 between 00:21 and 00:59

NY Lost Funds



JunK-Ythey will be bring back after the timeout
or use the returncallpark stuff
ya can redirect them to another place.
monstertruckreturncallpark?
JunK-Yyes
monstertruckcant find anything close to that in google..
`p4r14hcould anyone help me make Zaptel recognize a distinct ring, like the one for intercoms on a KSU from panasonic?
JunK-Ymonstertruck: wait
monstertruckif I can redirect the calls to another extension that would solve exactly the problem im having
lenne_dkThe asterisk flash panel can generate calls; click on a link, first your phone rings, when you pick up, the destination phone rings.
JunK-Ymonstertruck: http://bugs.digium.com/view.php?id=6953
with that, ya will be able to redirect the timeouted channel to antoher place
like to the reception or something.
lenne_dkOh, didn't notice it was so long (30 min) since you were talking about helpdesk calls...
monstertruckJunK-Y, awesome, thank you
JunK-Yif ya get any issue with that option, let me know.
monstertruckthat behavior of returning failed transfers to the caller is messing up my billing code
ok, will do
goldenearplease I woulk like to know : is this normal that when I receive a sip call the caller id is callername@myip ?
lokkju_wrkdepends on your config - normally, yes
JTdoesn't sound unusual
goldenearI would like it to be callername@callerdomain
otherwise I'm not able to call it back (missed call)
lokkju_wrk: really ? is it normal that the caller domain is stripped and replaced by the ip of my * server ?
yxaanyone knows what happened to chan_sccp?
goldenearag if 1.2.3.4 is the ip of my * server, then a incomming sip call from joe@sip.net is presented as joe@1.2.3.4 to my sip phone
ag/eg
JT is it the same for you ?
[TK]D-Fendergoldenear, Thats because * isn't a PROXY, its a B2BUA and ALL calls are as though coming direct from *
goldenear[TK]D-Fender: I see... that's why then. thanks for the explanation :)
but it's a big problem IMHO...
how can we call back a missed sip call then if we don't know the correct uri ?
is there a work around or something ?
JTgoldenear: most people don't get calls from random sip URIs
[TK]D-Fendergoldenear, This is kind of a rarity. Few people use * for direct un-authed SIP calls
goldenearbut what about ENUM ?
[TK]D-Fendergoldenear, Not relevent yet :)
Qwellyxa: It's dead
goldenear[TK]D-Fender: it is for me :)
already

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NY Lost Funds