Hey guys ,
i just got the asterisk forking again after giving up on the freepbx gui - i found that the config files were a little over complicated and slipt up…not sure yet if this wasthe right thing to do 
anyhow - i got the digium phones module working and if i give the phone an extension say 201 i get a dial tone and i can call other extensions
however i would like to have 3 buttons on the phone (D70) and each button picks a different sip account to dial out on - and eventually dahdi ) , but i can’t wrap my head about how to assign a button to a line - as when i create a line in res_digium_phone.conf, i end up getting silence in one case - or cannot dial any extensions in other
(note - i have no sip trunk configured yet - i want to get the internal calling working first)
Thank you !
Multiple lines are attached to Digium phones in res_digium_phone.conf by the use of multiple line= options in the type=phone section for the phone itself.
So, if you wanted line 201 on a phone it’d look like:
[myphone]
type=phone
line=201
...
If you wanted lines 201, 202 and 203 on a phone, it’d look like:
[myphone]
type=phone
line=201
line=202
line=203
when i did that i was not able to dial anything as i woudl get a fast busy
line was defined below in the same fine and would “extend” the same [201] from the sip.conf
is that right ?
how do i get it to “seize” the specific sip (or dahdi) for outbound ?
ok - i figured out a part of it
i did not have to resetup the “line” in the res_digium_phone.conf - now it works fine
well no gui approach is working well so far — i figured out the digitmaps to autodial - so moving ahead fast
Thank you!
how would i go about saying that if a user selects a “line” i want a specific dial plan
so in other words - you hit line 2 - i want to dial out on sip trunk 2
my extensions are in different “context” so can i use that to determine if dial plan is eligible ?
Each line / account on a phone has its own dial plan.
See the digit_map option - wiki.asterisk.org/wiki/display/ … ionOptions
Cheers
i understand the digit map to autodial after a number of coditions is met - but not how to dial from a dial plan
however, i think that the context is going to help me here from sip.conf…going to test it out … thanks
here is what i am runnign into
i set up skype sip in sip.conf
i set up a line in res_digium_phone.conf and add line to the phone
when i reload i get the following error
[Apr 30 14:31:03] ERROR[4333]: chan_sip.c:16832 register_verify: Peer ‘skype1’ is trying to register, but not configured as host=dynamic
[Apr 30 14:31:03] NOTICE[4333]: chan_sip.c:27786 handle_request_register: Registration from ‘“Skype1” sip:skype1@192.168.0.10’ failed for ‘192.168.0.10:5060’ - Peer is not supposed to register
if i configure multiple skype profiles only the last one reports this error
can you guide me as to where screwed up
Howdy,
If you’re doing a SIP peer that is Skype, you can’t use that same SIP peer on a phone.
See the error message:
[Apr 30 14:31:03] ERROR[4333]: chan_sip.c:16832 register_verify: Peer 'skype1' is trying to register, but not configured as host=dynamic
i need the phone line button to ring out on skype - how would you do that ?
You’d set up SIP peer for the line key on the phone, and then setup Asterisk dialplan to dial out that other SIP peer, that’s Skype.
ok - i kind of understand the reasoning behind that — so i did it … and after a few hiccups i got as far as this
No path to translate from SIP/skype1-0000000c to SIP/skypepeer-0000000b
(i called the line button skypepeer)
GOT IT … i googled the error - i had to match up the codecs
thank you malcomd - you pushed me and i got it working