OK, I am able to register devices and call anything within the internal context. But I can not dial a conference room. Can anyone that is able to dial a conference from an Asterisk instance please share their Sip.conf and Extension.conf so I can compare...?
On Wed, Aug 18, 2021 at 12:10 AM Maxim Solodovnik <[email protected]> wrote: > `sudo netstat -taupen|grep aster` > > lists port 5060 for me .... > > > On Wed, 18 Aug 2021 at 06:38, Yah's Global Kingdom <[email protected]> > wrote: > >> The SIP protocol uses port 5060, according to the documentation: SIP >> Config tcpenble =yes and tcpbindaddress default port number is 5060. >> >> On Mon, Aug 16, 2021 at 9:34 PM Maxim Solodovnik <[email protected]> >> wrote: >> >>> >>> >>> On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom <[email protected]> >>> wrote: >>> >>>> Please disregard, I have gotten the sip transport to enter the room. >>>> However, I don't see anything in Asterisk for when the Transport agent >>>> enters the room or when I try to register a client. >>>> >>> >>> You should "see something in Asterisk" at the moment the SIP user enters >>> the room (better with Om user in it ...) >>> >>> >>>> I have nothing listening on ports 5060,5061 or 5062. >>>> >>> >>> Why do you expect something should listen these ports? >>> >>> >>>> >>>> On Sat, Aug 14, 2021 at 11:29 AM Yah's Global Kingdom < >>>> [email protected]> wrote: >>>> >>>>> Update: >>>>> Asterisk is not listening on ports 5060/5061/5062 although I have >>>>> updated the sip.conf >>>>> >>>>> I am using the guide at >>>>> https://openmeetings.apache.org/AsteriskIntegration.html to implement >>>>> Asterisk and VOIP. >>>>> >>>>> Before under previous additions, when I entered the room, the SIP >>>>> transport agent would also enter the room. Now after upgrading from 5.0 >>>>> to >>>>> 6.10 when I enter the room no sip transport agent enters. What >>>>> information do I need to provide to anyone so I can troubleshoot this >>>>> matter. Is there an upgraded version of this guide >>>>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration >>>>> ? >>>>> The sipusers table in 6.1 looks nothing like the table in this guide. >>>>> >>>>> Sincerely >>>>> Bro Miles >>>>> YAH's Global Kingdom Ministries. >>>>> >>>>> On Sat, Aug 14, 2021 at 10:55 AM Yah's Global Kingdom < >>>>> [email protected]> wrote: >>>>> >>>>>> I am using the guide at >>>>>> https://openmeetings.apache.org/AsteriskIntegration.html to >>>>>> implement Asterisk and VOIP. >>>>>> >>>>>> Before under previous additions, when I entered the room, the SIP >>>>>> transport agent would also enter the room. Now after upgrading from 5.0 >>>>>> to >>>>>> 6.10 when I enter the room no sip transport agent enters. What >>>>>> information do I need to provide to anyone so I can troubleshoot this >>>>>> matter. Is there an upgraded version of this guide >>>>>> https://cwiki.apache.org/confluence/display/OPENMEETINGS/OpenMeetings+Asterisk+Integration >>>>>> ? The sipusers table in 6.1 looks nothing like the table in this guide. >>>>>> >>>>>> Sincerely >>>>>> Bro Miles >>>>>> YAH's Global Kingdom Ministries. >>>>>> >>>>> >>> >>> -- >>> Best regards, >>> Maxim >>> >> > > -- > Best regards, > Maxim >
