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Posted to user@openmeetings.apache.org by Keisuke Yamada <k-...@idnetworks.co.jp> on 2013/10/28 08:11:37 UTC
need some help for Integration with Asterisk
Hi
I am testing with asterisk right now.
http://openmeetings.apache.org/red5sip-integration_3.0.html
I did just like above.
When I tried to call 4001 , then I got a message like below.
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
-- Executing [4001@public:1] GotoIf("SIP/101-00000001", "0?ok:notavail")
in new stack
-- Goto (public,4001,7)
-- Executing [4001@public:7] Answer("SIP/101-00000001", "") in new stack
> 0x7f95d4000940 -- Probation passed - setting RTP source address to
192.168.0.150:5066
-- Executing [4001@public:8] Playback("SIP/101-00000001", "invalid") in
new stack
-- <SIP/101-00000001> Playing 'invalid.gsm' (language 'en')
> 0x7f95d4000940 -- Probation passed - setting RTP source address to
192.168.0.150:5066
> 0x7f95d4007620 -- Probation passed - setting RTP source address to
192.168.0.150:5068
-- Executing [4001@public:9] Hangup("SIP/101-00000001", "") in new stack
== Spawn extension (public, 4001, 9) exited non-zero on 'SIP/101-00000001'
So I saw "database show" on asterisk CLI, I could not see
"openmeetings/rooms/*".
I believe one of reason why it is failed is, Just asterisk did not
recognized Database for openmeetings.
So I tested "echo "select * FROM openmeetings.roomtype;" | isql -v
asterisk-connector".
I got a right response like, below
id | deleted| fieldvalues_id |name | starttime | updatetime |
| 1 | | 1541 | conference | 2013-10-24 11:37:35| |
| 2 | | -1 | audience | 2013-10-24 11:37:35| |
..
..
I think mysql's config is right.
I want to know how to check the configuration for this.
Please give me some advise.
Best Regards
Keisuke Yamada
|
Re: need some help for Integration with Asterisk
Posted by Alexei Fedotov <al...@gmail.com>.
Keisuke, I cannot see errors in the part of the log file you posted.
It seems all work as expected, doesn't it. This may worth examining
red5sip log.
I see some RTP traffic started to flow. This may be bold suggestion,
there may be a port problem.
Artyom, do you have any other suggestions?
--
With best regards / с наилучшими пожеланиями,
Alexei Fedotov / Алексей Федотов,
http://dataved.ru/
+7 916 562 8095
[1] Start using Apache Openmeetings today, http://openmeetings.apache.org/
[2] Join Alexei Fedotov @linkedin, http://ru.linkedin.com/in/dataved/
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On Mon, Oct 28, 2013 at 11:11 AM, Keisuke Yamada
<k-...@idnetworks.co.jp> wrote:
> Hi
>
> I am testing with asterisk right now.
>
> http://openmeetings.apache.org/red5sip-integration_3.0.html
> I did just like above.
>
> When I tried to call 4001 , then I got a message like below.
>
>
> == Using SIP VIDEO CoS mark 6
> == Using SIP RTP CoS mark 5
> -- Executing [4001@public:1] GotoIf("SIP/101-00000001", "0?ok:notavail")
> in new stack
> -- Goto (public,4001,7)
> -- Executing [4001@public:7] Answer("SIP/101-00000001", "") in new stack
>> 0x7f95d4000940 -- Probation passed - setting RTP source address to
> 192.168.0.150:5066
> -- Executing [4001@public:8] Playback("SIP/101-00000001", "invalid") in
> new stack
> -- <SIP/101-00000001> Playing 'invalid.gsm' (language 'en')
>> 0x7f95d4000940 -- Probation passed - setting RTP source address to
> 192.168.0.150:5066
>> 0x7f95d4007620 -- Probation passed - setting RTP source address to
> 192.168.0.150:5068
> -- Executing [4001@public:9] Hangup("SIP/101-00000001", "") in new stack
> == Spawn extension (public, 4001, 9) exited non-zero on 'SIP/101-00000001'
>
> So I saw "database show" on asterisk CLI, I could not see
> "openmeetings/rooms/*".
> I believe one of reason why it is failed is, Just asterisk did not
> recognized Database for openmeetings.
>
> So I tested "echo "select * FROM openmeetings.roomtype;" | isql -v
> asterisk-connector".
> I got a right response like, below
>
> id | deleted| fieldvalues_id |name | starttime | updatetime |
> | 1 | | 1541 | conference | 2013-10-24 11:37:35| |
> | 2 | | -1 | audience | 2013-10-24 11:37:35| |
> ..
> ..
>
> I think mysql's config is right.
>
> I want to know how to check the configuration for this.
> Please give me some advise.
>
> Best Regards
> Keisuke Yamada
> |
Re: need some help for Integration with Asterisk
Posted by Keisuke Yamada <k-...@idnetworks.co.jp>.
Hi Maxim
Thank you for your reply.
Set(CONFBRIDGE(user,pin)=1)
means set pin as 1, so if you press 1 it was ok.
Anyway I will check about Asterisk, now.
Thank you.
Keisuke Yamada
(2013/10/29 12:08), Maxim Solodovnik wrote:
> Unfortunately I'm not very good in Asterisk configuration
> Pin can be entered in Administration->Conference Rooms section
>
>
> On Tue, Oct 29, 2013 at 9:00 AM, Keisuke Yamada
> <k-yamada@idnetworks.co.jp <ma...@idnetworks.co.jp>> wrote:
>
> Hi
>
> I guessed that there is very few information for DB(openmeetings/room)
> on asterisk.
>
> so I rewrite dailplan like below
>
> [rooms2]
> exten => _500X!,1,Set(CONFBRIDGE(user,template)=sip_user)
> exten => _500X!,n,Set(CONFBRIDGE(user,pin)=1)
> exten => _500X!,n,Confbridge(${EXTEN},default_bridge,)
> exten => _500X!,n,Hangup
>
> when I test to dial 5001, system says "please enter pin number".
> Is there some one who knows about pin number for conference room with
> asterisk.
>
> Best Regards
> Keisuke Yamada
>
>
> (2013/10/28 16:11), Keisuke Yamada wrote:
> > Hi
> >
> > I am testing with asterisk right now.
> >
> > http://openmeetings.apache.org/red5sip-integration_3.0.html
> > I did just like above.
> >
> > When I tried to call 4001 , then I got a message like below.
> >
> >
> > == Using SIP VIDEO CoS mark 6
> > == Using SIP RTP CoS mark 5
> > -- Executing [4001@public:1] GotoIf("SIP/101-00000001",
> "0?ok:notavail")
> > in new stack
> > -- Goto (public,4001,7)
> > -- Executing [4001@public:7] Answer("SIP/101-00000001", "") in
> new stack
> >> 0x7f95d4000940 -- Probation passed - setting RTP source address to
> > 192.168.0.150:5066 <http://192.168.0.150:5066>
> > -- Executing [4001@public:8] Playback("SIP/101-00000001",
> "invalid") in
> > new stack
> > -- <SIP/101-00000001> Playing 'invalid.gsm' (language 'en')
> >> 0x7f95d4000940 -- Probation passed - setting RTP source address to
> > 192.168.0.150:5066 <http://192.168.0.150:5066>
> >> 0x7f95d4007620 -- Probation passed - setting RTP source address to
> > 192.168.0.150:5068 <http://192.168.0.150:5068>
> > -- Executing [4001@public:9] Hangup("SIP/101-00000001", "") in
> new stack
> > == Spawn extension (public, 4001, 9) exited non-zero on
> 'SIP/101-00000001'
> >
> > So I saw "database show" on asterisk CLI, I could not see
> > "openmeetings/rooms/*".
> > I believe one of reason why it is failed is, Just asterisk did not
> > recognized Database for openmeetings.
> >
> > So I tested "echo "select * FROM openmeetings.roomtype;" | isql -v
> > asterisk-connector".
> > I got a right response like, below
> >
> > id | deleted| fieldvalues_id |name | starttime | updatetime |
> > | 1 | | 1541 | conference | 2013-10-24 11:37:35| |
> > | 2 | | -1 | audience | 2013-10-24 11:37:35| |
> > ..
> > ..
> >
> > I think mysql's config is right.
> >
> > I want to know how to check the configuration for this.
> > Please give me some advise.
> >
> > Best Regards
> > Keisuke Yamada
> > |
> >
>
>
>
>
> --
> WBR
> Maxim aka solomax
Re: need some help for Integration with Asterisk
Posted by Maxim Solodovnik <so...@gmail.com>.
Unfortunately I'm not very good in Asterisk configuration
Pin can be entered in Administration->Conference Rooms section
On Tue, Oct 29, 2013 at 9:00 AM, Keisuke Yamada
<k-...@idnetworks.co.jp>wrote:
> Hi
>
> I guessed that there is very few information for DB(openmeetings/room)
> on asterisk.
>
> so I rewrite dailplan like below
>
> [rooms2]
> exten => _500X!,1,Set(CONFBRIDGE(user,template)=sip_user)
> exten => _500X!,n,Set(CONFBRIDGE(user,pin)=1)
> exten => _500X!,n,Confbridge(${EXTEN},default_bridge,)
> exten => _500X!,n,Hangup
>
> when I test to dial 5001, system says "please enter pin number".
> Is there some one who knows about pin number for conference room with
> asterisk.
>
> Best Regards
> Keisuke Yamada
>
>
> (2013/10/28 16:11), Keisuke Yamada wrote:
> > Hi
> >
> > I am testing with asterisk right now.
> >
> > http://openmeetings.apache.org/red5sip-integration_3.0.html
> > I did just like above.
> >
> > When I tried to call 4001 , then I got a message like below.
> >
> >
> > == Using SIP VIDEO CoS mark 6
> > == Using SIP RTP CoS mark 5
> > -- Executing [4001@public:1] GotoIf("SIP/101-00000001", "0?ok:notavail")
> > in new stack
> > -- Goto (public,4001,7)
> > -- Executing [4001@public:7] Answer("SIP/101-00000001", "") in new stack
> >> 0x7f95d4000940 -- Probation passed - setting RTP source address to
> > 192.168.0.150:5066
> > -- Executing [4001@public:8] Playback("SIP/101-00000001", "invalid") in
> > new stack
> > -- <SIP/101-00000001> Playing 'invalid.gsm' (language 'en')
> >> 0x7f95d4000940 -- Probation passed - setting RTP source address to
> > 192.168.0.150:5066
> >> 0x7f95d4007620 -- Probation passed - setting RTP source address to
> > 192.168.0.150:5068
> > -- Executing [4001@public:9] Hangup("SIP/101-00000001", "") in new stack
> > == Spawn extension (public, 4001, 9) exited non-zero on
> 'SIP/101-00000001'
> >
> > So I saw "database show" on asterisk CLI, I could not see
> > "openmeetings/rooms/*".
> > I believe one of reason why it is failed is, Just asterisk did not
> > recognized Database for openmeetings.
> >
> > So I tested "echo "select * FROM openmeetings.roomtype;" | isql -v
> > asterisk-connector".
> > I got a right response like, below
> >
> > id | deleted| fieldvalues_id |name | starttime | updatetime |
> > | 1 | | 1541 | conference | 2013-10-24 11:37:35| |
> > | 2 | | -1 | audience | 2013-10-24 11:37:35| |
> > ..
> > ..
> >
> > I think mysql's config is right.
> >
> > I want to know how to check the configuration for this.
> > Please give me some advise.
> >
> > Best Regards
> > Keisuke Yamada
> > |
> >
>
>
--
WBR
Maxim aka solomax
Re: need some help for Integration with Asterisk
Posted by Keisuke Yamada <k-...@idnetworks.co.jp>.
Hi
I guessed that there is very few information for DB(openmeetings/room)
on asterisk.
so I rewrite dailplan like below
[rooms2]
exten => _500X!,1,Set(CONFBRIDGE(user,template)=sip_user)
exten => _500X!,n,Set(CONFBRIDGE(user,pin)=1)
exten => _500X!,n,Confbridge(${EXTEN},default_bridge,)
exten => _500X!,n,Hangup
when I test to dial 5001, system says "please enter pin number".
Is there some one who knows about pin number for conference room with
asterisk.
Best Regards
Keisuke Yamada
(2013/10/28 16:11), Keisuke Yamada wrote:
> Hi
>
> I am testing with asterisk right now.
>
> http://openmeetings.apache.org/red5sip-integration_3.0.html
> I did just like above.
>
> When I tried to call 4001 , then I got a message like below.
>
>
> == Using SIP VIDEO CoS mark 6
> == Using SIP RTP CoS mark 5
> -- Executing [4001@public:1] GotoIf("SIP/101-00000001", "0?ok:notavail")
> in new stack
> -- Goto (public,4001,7)
> -- Executing [4001@public:7] Answer("SIP/101-00000001", "") in new stack
>> 0x7f95d4000940 -- Probation passed - setting RTP source address to
> 192.168.0.150:5066
> -- Executing [4001@public:8] Playback("SIP/101-00000001", "invalid") in
> new stack
> -- <SIP/101-00000001> Playing 'invalid.gsm' (language 'en')
>> 0x7f95d4000940 -- Probation passed - setting RTP source address to
> 192.168.0.150:5066
>> 0x7f95d4007620 -- Probation passed - setting RTP source address to
> 192.168.0.150:5068
> -- Executing [4001@public:9] Hangup("SIP/101-00000001", "") in new stack
> == Spawn extension (public, 4001, 9) exited non-zero on 'SIP/101-00000001'
>
> So I saw "database show" on asterisk CLI, I could not see
> "openmeetings/rooms/*".
> I believe one of reason why it is failed is, Just asterisk did not
> recognized Database for openmeetings.
>
> So I tested "echo "select * FROM openmeetings.roomtype;" | isql -v
> asterisk-connector".
> I got a right response like, below
>
> id | deleted| fieldvalues_id |name | starttime | updatetime |
> | 1 | | 1541 | conference | 2013-10-24 11:37:35| |
> | 2 | | -1 | audience | 2013-10-24 11:37:35| |
> ..
> ..
>
> I think mysql's config is right.
>
> I want to know how to check the configuration for this.
> Please give me some advise.
>
> Best Regards
> Keisuke Yamada
> |
>