Re: RES: RES: [Asterisk-Users] 403 Forbidden

2004-03-12 Thread Mireia Munoz de jesus
The codecs are: SIP Phone: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 Asterisk: in sip.conf 1: ulaw 2: alaw in oh323.conf 1: G711U Gateway: preference 1: G711U preference 2: . . . preference 8: G711A That's good? Can you see where's the

Re: RES: RES: [Asterisk-Users] 403 Forbidden

2004-03-12 Thread Michael Manousos
Mireia Munoz de jesus wrote: The codecs are: SIP Phone: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 Asterisk: in sip.conf 1: ulaw 2: alaw in oh323.conf 1: G711U Gateway: preference 1: G711U preference 2: . . . preference 8: G711A Try with

RES: RES: [Asterisk-Users] 403 Forbidden

2004-03-11 Thread Vinicius Viana
The call end reason EndedByQ931Cause is used by the OpenH323 stack when it doesn't know the real cause. Try to see if the codecs in the gateway are compatible with the codecs in asterisk. What are the codecs you are using in SIP Phones, in Asterisk and in the gateway? Regards, Vinicius