On Sun, 22 Aug 2021 at 01:06, Yah's Global Kingdom <[email protected]> wrote:
> 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...? > I was able to dial the room using a softphone (Linphone) ... The configs are at the main site :) > 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 >> > -- Best regards, Maxim
