On Sat, Dec 17, 2005 at 07:44:29AM -0600, Rich Adamson wrote: ok rick all of my conf... asterisk 1.2.1 zaptel 1.2.1
i have a pbx simple with digital phones in one side. and the other side are xten with SIP. my extencion.conf [general] static=yes writeprotect=no autofallthrough=yes [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=Zap/g1 [local] ; ignorepat => 9 include => default [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. exten => 402,1,Dial(SIP/402,20) exten => 402,2,Hangup [teste] exten => s,1,Dial(SIP/402,20) exten => s,2,Hangup exten => 402,1,Dial(SIP/402,20) exten => 402,2,Hangup exten => _XXX,1,Dial(${TRUNK}/${EXTEN}) exten => _XXX,2,Voicemail(u${EXTEN}) the sip.conf is the default for asterisk i didnt touch anything in this file only the extention number and i dont have nothing about codecs in this file [402] type=friend host=dynamic username=Pablo secret=teste callerid="Pablo" <402> canreinvite=no ;nat=yes ;amaflags=billing context=teste > > > > Hi all i have some problems with my pbx and asterisk codecs. > > > > > > > > if i use g711u or g711a codecs. the line never hangup. and the origin > > > > and destination are connected until i restart my pbx or asterisk > > > > > > > > But if i use GSM all work fine. > > > > > > > > is possible to solve this problem? or use only gsm codec? > > > > > > > > Yes, its possible to solve the problem. > > > > can you explain how? > > Not without you providing at least "something" to give us a clue what it > is that you've programmed into your system. > > How about if you give us some clue as to which version of * you're > using, what type of phones are associated with "origin" and "destination", > if these are sip phones what do your sip.conf definitions look like, > what does the appropriate sections of extensions.conf look like, and > any other configuration pieces that might pertain to whatever it is > that you've implemented. Your posting implies there might be more than > one * system involved and possibly even iax trunking, etc. > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---end quoted text--- -- .- Pablo Allietti LACNIC _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users