Re: [Asterisk-Users] SIP-H.323 dial tone and busy tone problem.
Try playing with faststart . Moises Silva wrote: Hi Carlos. I have never used H323. But im interested in your problem. Have you tried to use de 'sip debug' 'iax2 debug' commands? and check the console with a high verbosity level? could you post any warning or relevant output when the call is made? best regards On 6/11/05, Carlos Alberto Lara de Hoyos [EMAIL PROTECTED] wrote: Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-asteriskh323-g729--nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone Network's Open H.323 Channel Driver G.729/PCM16 Codec Translator Raw G729 data It is a problem of codecs compatiblility or wath? Thanks to all. ___ 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 -- Best Regards, Damian Minkov COSMOS Software Enterprises, Ltd. SIP:[EMAIL PROTECTED] Tel:(+359-2) 856-19-43 400-55-65 Mobile: (+359-88) 853-28-25 E-Mail: [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 3, fl. 1, entr. A, Preki pat str., No. 16, kv. Pavlovo, 1618 Sofia, Bulgaria ___ 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] SIP-H.323 dial tone and busy tone problem.
Hi Carlos. I have never used H323. But im interested in your problem. Have you tried to use de 'sip debug' 'iax2 debug' commands? and check the console with a high verbosity level? could you post any warning or relevant output when the call is made? best regards On 6/11/05, Carlos Alberto Lara de Hoyos [EMAIL PROTECTED] wrote: Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-asteriskh323-g729--nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone Network's Open H.323 Channel Driver G.729/PCM16 Codec Translator Raw G729 data It is a problem of codecs compatiblility or wath? Thanks to all. ___ 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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ 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] SIP-H.323 dial tone and busy tone problem.
Greetings to the list: this is my problen when I make a call from my asterisk towards a nortel PBX , the call is made but in my telephone sip I do not listen the dial tone or the busy tone but the call it is completed normally. sip-phone-g729-asteriskh323-g729--nortel-pbx thi is may configuration: RedHat 8 2.4.18-14 Asterisk 1.0.7 The NuFone Network's Open H.323 Channel Driver G.729/PCM16 Codec Translator Raw G729 data It is a problem of codecs compatiblility or wath? Thanks to all. ___ 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
