search for: grandstreem

I've only gotten to hear Music On Hold once. I am running the right version of mpg123 .059r as downloaded from mpg123.de and compiled locally So I'm looking for any help on getting this up and running. I can see on the console that the SIP phone is placing the call on hold, but there is no audio coming out on the other phone. Mucho, thanks john

Hi list, I've got a testbed running with the following config: 1. RH 7.3 linux machine 2. 2 Grandstream phones 3. 2 XTen soft clients When I dial voice mail I have a problem. here is the flow. 1. dial 8500 (the exten for voice mail) 2. enter 2600 via touch pad on the grandstreem 3. entire passwd via touch pad I always get an invalid login When looking at the Ast console I see that the exten number received by Ast is 260 not 2600. This happens no matter how fast or slow (delay between pushing keys on phone) things happen. Any ideas ?? thanks mucho john

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the