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A maintained and still updated list archive can be found at https://lists.osmocom.org/hyperkitty/list/OpenBSC@lists.osmocom.org/.
Nik Pakar nikpakar at gmail.comHi 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 at 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 > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.osmocom.org/pipermail/openbsc/attachments/20120626/c498ebc2/attachment.htm>