Hi again... normally the 0/16 is a d-channel.
check the config in the zapata.conf. You should have some thing like this /etc/zapata.conf bchan=1-15 dchan=16 bchan=17-31 /etc/asterisk/zapata.conf channel => 1-15,17-31 i don't rember exactelly but in /proc/zaptel there is the possibility to check if the channels are in use or not. Maybe someone else can give you a hint... sorry but i only interconnect Asterisk and H4K using chan_capi and i have no experience with zapata ;-( rich --- Josué Conti <[EMAIL PROTECTED]> wrote: > Jun 26 12:43:16 WARNING[31148]: chan_zap.c:8386 > pri_dchannel: Ring requested > on unconfigured channel 0/16 span 1 > I noticed this message in the CLI, when I tried to > effect one call of HiPath > 4000 for asterisk. Ring occurred, however when the > voicemail of asterisk > took care of call it was dumb, without no sound. I > thank the attention > Regards > > Josué > > 2006/6/26, Josué Conti <[EMAIL PROTECTED]>: > > > > Hi Richard. > > Thank you very much for its attention. In the > reality what is occurring is > > that in some originated calls of the HiPath with > destination to the Asterisk > > they are being without the dumb and rings. I do > not have this parameter in > > my HiPath 4000, what I have seemed in the COT is > TR6T (1tr6 isdn tie link) > > would be this parameter? > Best Regards > > Josué > > > > 2006/6/26, richard Coco <[EMAIL PROTECTED]>: > > > > > > > > Hi Josué > > > > > > if the Siemens phone calls Asterisk, it didn't > get a > > > dial tone from Asterisk? Is it correct? > > > > > > if yes, this is depending of Asterisk which > didn't > > > generates a ringback messages as it expexts dial > ton > > > generation localy. So try this workaround for > HiPath > > > local dial ton generation: > > > -> Add option TR6Q(TRGT) to the class of trunk > (COT) > > > parameters > > > > > > hope it will help... > > > > > > rich > > > > > > > > > > > > > > > > > > --- Josué Conti <[EMAIL PROTECTED]> wrote: > > > > > > > Hello all. > > > > I have installed and functioning > asterisk-1.2.9.1 > > > > where I effected one > > > > upgrade in asterisk-1.0.9, is interconnected > with a > > > > PABX Siemens HiPath 4000 > > > > in ISDN PRI with protocol QSIG, the one that > is > > > > happening he is that the > > > > calls originated for PABX Siemens and destined > to > > > > SIP phones asterisk are > > > > being without audio, nor Ring, is dumb. They > could > > > > help in this case me? > > > > Best Regards > > > > > > > > Josué > > > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by > Easynews.com<http://easynews.com/> > > > > -- > > > > > > > > Asterisk-Users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > > > > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! Mail has the best spam > protection around > > > http://mail.yahoo.com > > > _______________________________________________ > > > --Bandwidth and Colocation provided by > Easynews.com<http://easynews.com/>-- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > _______________________________________________ > --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 > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ --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
