This is merely a historical archive of years 2008-2021, before the migration to mailman3.
A maintained and still updated list archive can be found at https://lists.osmocom.org/hyperkitty/list/OpenBSC@lists.osmocom.org/.
Ellen Apolinar ellen.apolinar.td at googlemail.comHey Don, you are great! Thanks for your help, it worked. I can load hfcmulti and it is list under <lspci -v> for the HFC-E1 card and also <misdn_info> founds the card. I'm very happy that it is working again after it broke. :-) But the problem with lcr is still there: > <lcr start> > > ** LCR Version 1.7 > > 000000 DEBUG (in getrulesetbyname() line 1871): ruleset main found. > GSM is enabled, but not compiled. Use --with-gsm-bs or --with-gsm-ms while > configure! > --------------------- 2012.06.25 13:45:11 000001 > 000001 DEBUG (in join_free() line 99): no more pending join(s), done! > > And <lcradmin state> shows: > Failed to connect to socket "/var/tmp/LCR.socket". > Is LCR running? > @Andreas: Thanks, I didn't know that. I prefer to work with mISDN & LCR & Asterisk because I have these installed and if it is possible to run OpenBSC with mISDN, mISDNuser, lcr and Asterisk I want to try this. Do you know if it is required to use lcr with gsm for connect OpenBSC and Asterisk if I don't use sip? Regards and thank you for your help. Ellen 2012/6/24 Andreas Eversberg <andreas at eversberg.eu> > 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/20120625/cd4a7cd7/attachment.htm>