RE : [Asterisk-Users] Unable to create channel of type 'Zap'
Hi there I have troubles while installing oh323. I got from cvs the asterisk_oh323, and then I tried to make install, in the end I have this: ... /usr/include/ptlib/array.h:341: `dynamic' undeclared (first use this function) /usr/include/ptlib/array.h:341: only constructors take base initializers /usr/include/ptlib/array.h:341: confused by earlier errors, bailing out make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323/wrapper' make: *** [subdirs_all] Error 1 Could you plese illuminate the way for me, I´ll be thankfull. May thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No ring tone while calling from h323 to SIP
I set up asterisk since a while, I had no problem using it, but now I just added H323 module, it´s working fine, but when I place calls from H323 to pstn via Asterisk ; idon´t hear the ring tone, it goes silent until the other part pickup the phone and we can talk. Could someone help me please. Thank you and good luck to everybody. Jalil. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] sound problem
Check the used codecs, if it´s according to the codec used by your sip users. Welcome -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Muhammad Rizwan Khan Envoyé : lundi 10 janvier 2005 18:53 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] sound problem Is there any config file related to voice, which should be change in order to hear the sound in dialer? On Monday 10 January 2005 21:23, you wrote: > I have configured asterisk, but when i calls from my dialler, it connects > successfully, but did not give any voice at both ends. > What should i need to do? > > Thanks > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
