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Hi It seems, you should set up 2 different parts of your system: 1. Sync master/slave separation; 2. Telephony signalling master/slave separation. Since the VoIP trunks are fundamentally asyncronous, then SIP gateway should take sync from md110. It's absolutely mandatory. See Quintum manual...

Hi all. I got a problem. Bc12sp11 had digital extension #1213, that had to be transformed to external no. There was some mdns, presented on 1213. These mdns wasn't removed before ksexe:dir=1213. Usually this procedure goes correctly. But in my case, I guess, one of mdns, 1271, was in speech...

does anyone have experience in fine tuning of codecs interworking in bc12 and cisco? i got only signalling negotiation: calls passes to both md110 sides, but no speech. any ideas?

Hi. Not a secret. Examples: 1.User leaved his office, but wants to remain his number the same. md can't reach his new point. so i have to receive his old number from pstn, convert it into new and transfer the call back to pstn by the same path. 2.MD is transit pbx between pstn and other small...

Does anybody have a bc10 patch to return an incoming call to rou from which it came (isdn pri)? Usually this causes congestion, but i know, that pscasi exists for bc9 on cmps5 unit. And it works. Any ideas?

Look here: http://www.cisco.com/univercd/cc/td/doc/product/access/acs_mod/cis2600/sw_conf/pbx_int/ericsson.htm Maybe u'll find complete answer.