|JunK-Y||they will be bring back after the timeout|
or use the returncallpark stuff
ya can redirect them to another place.
|monstertruck||cant find anything close to that in google..|
|`p4r14h||could anyone help me make Zaptel recognize a distinct ring, like the one for intercoms on a KSU from panasonic?|
|monstertruck||if I can redirect the calls to another extension that would solve exactly the problem im having|
|lenne_dk||The asterisk flash panel can generate calls; click on a link, first your phone rings, when you pick up, the destination phone rings.|
with that, ya will be able to redirect the timeouted channel to antoher place
like to the reception or something.
|lenne_dk||Oh, didn't notice it was so long (30 min) since you were talking about helpdesk calls...|
|monstertruck||JunK-Y, awesome, thank you|
|JunK-Y||if ya get any issue with that option, let me know.|
|monstertruck||that behavior of returning failed transfers to the caller is messing up my billing code|
ok, will do
|goldenear||please I woulk like to know : is this normal that when I receive a sip call the caller id is callername@myip ?|
|lokkju_wrk||depends on your config - normally, yes|
|JT||doesn't sound unusual|
|goldenear||I 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 ?
|yxa||anyone knows what happened to chan_sccp?|
|goldenear||ag if 220.127.116.11 is the ip of my * server, then a incomming sip call from firstname.lastname@example.org is presented as email@example.com to my sip phone|
JT is it the same for you ?
|[TK]D-Fender||goldenear, 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 ?
|JT||goldenear: most people don't get calls from random sip URIs|
|[TK]D-Fender||goldenear, This is kind of a rarity. Few people use * for direct un-authed SIP calls|
|goldenear||but what about ENUM ?|
|[TK]D-Fender||goldenear, Not relevent yet :)|
|Qwell||yxa: It's dead|
|goldenear||[TK]D-Fender: it is for me :)|