#asterisk - Mon 16 Apr 2007 between 14:47 and 15:04



actionblitzrage can't wait for his E61i to arrive
blitzragecan't wait for his E61i to arrive
Mercestesizaak, Then you can exten => _400x,1,Dial(SIP/${EXTEN},180)
Qwell[]blitzrage: chan_cellphone!
blitzrageQwell[]: exactly!!!
Qwell[]it's the hotness
blitzrageand I can run a SIP client on it too
Mercestesizaak, Or more appropriately, exten => _400[1-5],1,Dial(STIP,${EXTEN},180)
blitzragenow I need a bluetooth USB adapter for my computer though
Qwell[]blitzrage: They can be had for $20ish
blitzragedon't think this laptop has bluetooth...
Qwell[]meh, I'm an idiot
blitzrageyah, I imagine they are fairly inexpensive
ok :)
Qwell[]I decided it wasn't worth the $15 to add bluetooth to my laptop
would've been internal and everything, but no...
izaakMercestes: thanks, i think i will use that sort of pattern for incoming. but for outgoing, where i need to dial different IAX channels depending on which SIP channel?
Qwell[]now if I want it, it's like $40, heh
blitzrageheh
that was kinda dumb :)
Qwell[]totally
oh well
Mercestesizaak, Are you trying to recreate static lines for you rphones using IAX?
Qwell[]I also only got the 40gb hd...
which was also an incredibly dumb thing to do
now the $40 I saved is gonna cost me like $120 =x
izaakMercestes: i'm not sure what you mean. my VOIP provider has given me an IAX channel per DID for incoming and outgoing. for incoming i understand how to use just one context. but i'm confused about outgoing.
Mercestes: each phone on my network has its own DID
Mercestesizaak, Yo ushould be able to dial out dynamically.
NLokhi does anyone have experience connecting Asterisk to VoiceGenie? I am having problem transfering a call.
[TK]D-FenderNLok: just describe the problem you're having. Pastbin CLI output with SIP debug enabled where applicable.
~pb
jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste
ManxPowerurpmi /tmp/mpeg4ip-1.5.0.1/player/src/video_sdl.cpp:280: undefined reference to `XMoveWindow'
drat
Qwell[]ManxPower: silly manduck
tzafrirkhronos, http://lists.digium.com/pipermail/asterisk-users/2007-April/184970.html
zaidehi
anyone have an idea to select my second VOIP provider when the first provider is out or timeout (or others errors)? i haven't found how to do it in my extension.conf
Qwell[]zaide: after Dial, check the value of the DIALSTATUS variable
LeddyHMAny thoughts on an os migration through vmware for ~30 users? "It will be fine, should be ok, audio is too choppy, no worries it's fine" for about a week?

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