Re: A ramble about MSC, sip-connector and asterisk.

Hi Gullik, what is up with your email client? It seems to insert two line feeds every 100 characters or so. I've never seen such weird flow as in your mails. Harder to follow your sentences when they break in the middle. Using external PBX is not trivial, and as soon as you do, all local call switching in osmo-msc is switched off. In general, using osmo-sip-connector works; what is not satisfactory yet is the codec negotiation. Sadly, so far the easiest solution is to hardcode a payload type number into osmo-sip-connector and clamp everything to one specific codec. For example, look at the neels/fr branch in osmo-sip-connector.git. But yours sounds more like a SIP negotiation issue?? I haven't used asterisk yet, but AFAIK at least yate, freeswitch and kamailio work without "clever hacking". Nah. osmo-bts *never* bridges nor ever bridged calls. Old openbsc's osmo-nitb? openbts?? Since you mentioned lime before, let me say that even using timing-accurate sysmoBTS, I had 90% handover failure until I properly calibrated the internal clock. With a lime and no external clock source, you're hopelessly doomed. ~N