`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
