Hi Andreas,

Do we still have to check out from jolly-rtp branch or the rtp-bridge now work with main openbsc repo ?

Rgds
Nik

On Sun, Jun 24, 2012 at 12:13 PM, Andreas Eversberg <andreas@eversberg.eu> wrote:
Ellen Apolinar wrote:

OpenBSC seems to work without errors but to connect it with asterisk I need mISDN, mISDNuser and LCR.

hi,

if you like to use lcr with gsm (bs or ms), then you cannot use asterisk channel driver. it only works with isdn. but you can use sip. in order to do that you may:

- disable mISDN  (--without-misdn)
- enable sip (--with-sip), you also need to have sipsofia installed
- add a sip interface (see default/interface.conf). then everything is possible without mISDN, but you cannot use isdn phones/lines in this setup.

example to just connect GSM and SIP interface without routing:

 [GSM]
gsm-bs
tones yes
earlyb no
#rtp-bridge
bridge SIP


[SIP]
sip <local ip>[:local sip port] <asterisk sip ip>[:asterisk sip port]
tones no
earlyb yes
#rtp-bridge
bridge GSM

if asterisk and lcr run on the same machine, you need to change the sip port on asterisk or lcr side.

if you like to use GSM codec on asterisk side, you may enable rtp-bridge. then the asterisk directly negotiates the codec with the phone.
in this case GSM codec must be supported by asterisk. tested codecs are FR(standard) and EFR.

regards,

andreas