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Posted to user@openmeetings.apache.org by Yah's Global Kingdom <ya...@gmail.com> on 2021/08/14 17:55:07 UTC
SIP Integration
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.
Re: SIP Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
On Sun, 22 Aug 2021 at 01:06, Yah's Global Kingdom <ya...@gmail.com>
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 <so...@gmail.com>
> wrote:
>
>> `sudo netstat -taupen|grep aster`
>>
>> lists port 5060 for me ....
>>
>>
>> On Wed, 18 Aug 2021 at 06:38, Yah's Global Kingdom <ya...@gmail.com>
>> 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 <so...@gmail.com>
>>> wrote:
>>>
>>>>
>>>>
>>>> On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom <ya...@gmail.com>
>>>> 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 <
>>>>> yahsgkm@gmail.com> 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 <
>>>>>> yahsgkm@gmail.com> 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
Re: SIP Integration
Posted by Yah's Global Kingdom <ya...@gmail.com>.
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 <so...@gmail.com>
wrote:
> `sudo netstat -taupen|grep aster`
>
> lists port 5060 for me ....
>
>
> On Wed, 18 Aug 2021 at 06:38, Yah's Global Kingdom <ya...@gmail.com>
> 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 <so...@gmail.com>
>> wrote:
>>
>>>
>>>
>>> On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom <ya...@gmail.com>
>>> 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 <
>>>> yahsgkm@gmail.com> 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 <
>>>>> yahsgkm@gmail.com> 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
>
Re: SIP Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
`sudo netstat -taupen|grep aster`
lists port 5060 for me ....
On Wed, 18 Aug 2021 at 06:38, Yah's Global Kingdom <ya...@gmail.com>
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 <so...@gmail.com>
> wrote:
>
>>
>>
>> On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom <ya...@gmail.com>
>> 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 <ya...@gmail.com>
>>> 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 <
>>>> yahsgkm@gmail.com> 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
Re: SIP Integration
Posted by Yah's Global Kingdom <ya...@gmail.com>.
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 <so...@gmail.com>
wrote:
>
>
> On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom <ya...@gmail.com>
> 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 <ya...@gmail.com>
>> 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 <ya...@gmail.com>
>>> 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
>
Re: SIP Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
On Mon, 16 Aug 2021 at 21:55, Yah's Global Kingdom <ya...@gmail.com>
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 <ya...@gmail.com>
> 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 <ya...@gmail.com>
>> 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
Re: SIP Integration
Posted by Yah's Global Kingdom <ya...@gmail.com>.
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. I have nothing
listening on ports 5060,5061 or 5062.
On Sat, Aug 14, 2021 at 11:29 AM Yah's Global Kingdom <ya...@gmail.com>
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 <ya...@gmail.com>
> 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.
>>
>
Re: SIP Integration
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello,
I would use this https://openmeetings.apache.org/AsteriskIntegration.html
guide (I followed it while I wrote it)
The guide at Confluence should be outdated, please ignore it
On Sun, 15 Aug 2021 at 01:29, Yah's Global Kingdom <ya...@gmail.com>
wrote:
> Update:
> Asterisk is not listening on ports 5060/5061/5062 although I have updated
> the sip.conf
>
I'm not sure why you need this port :(
it isn't mentioned at
https://openmeetings.apache.org/AsteriskIntegration.html
> 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.
>
I would double-check all steps
then share the logs (both Asterisk debug logs and OM logs)
> 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 <ya...@gmail.com>
> 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
Re: SIP Integration
Posted by Yah's Global Kingdom <ya...@gmail.com>.
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 <ya...@gmail.com>
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.
>