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mkezys at gmail.com
Guest
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 Posted: Mon Jun 02, 2008 6:11 am Post subject: [asterisk-users] Realtime SIP registration for multiple Aste |
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Hello,
We are building solution from multiple Asterisk servers. They are using
Realtime. Servers can be added dynamically over GUI.
Each server can register to different provider/carrier over SIP.
Question would be: how to make it work using pure Realtime - without editing
.conf files?
We can't use same sip.conf in the Realtime database because servers register
to different provider/carriers and for other server-specific options
(externip/etc).
That could be possible if only 1 Aserisk server would be in the system. What
to do with many Asterisk servers?
Is here any workaround to get registration work with different servers using
only Realtime?
Thank you.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com |
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atis at iq-labs.net
Guest
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| Posted: Mon Jun 02, 2008 6:53 am Post subject: [asterisk-users] Realtime SIP registration for multiple Aste |
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On Mon, Jun 2, 2008 at 2:11 PM, Mindaugas Kezys <mkezys at gmail.com> wrote:
| Quote: | Hello,
We are building solution from multiple Asterisk servers. They are using
Realtime. Servers can be added dynamically over GUI.
Each server can register to different provider/carrier over SIP.
Question would be: how to make it work using pure Realtime - without editing
.conf files?
We can't use same sip.conf in the Realtime database because servers register
to different provider/carriers and for other server-specific options
(externip/etc).
That could be possible if only 1 Aserisk server would be in the system. What
to do with many Asterisk servers?
Is here any workaround to get registration work with different servers using
only Realtime?
|
Hi Mindaugas,
well, unfortunately there is no way how to add "register" entries in
realtime. Main problem behind this is that those entries are read on
asterisk startup and if You would want to add some peer dynamically,
Asterisk would need to get notified about this.
Only solution that could work is generating "register =>" entries from
database by custom script in "#exec" (i suppose You have to enable
execincludes in asterisk.conf). So, upon start or reload of chan_sip
asterisk will execute your script, which would return correct
providers. So, after adding provider dynamically, You would need to
reload chan_sip.
If you would be interested in developing some kind of mechanism to
allow reading "register" entries from realtime, I could assist You in
some way, as this would be useful to me too, but specification of this
should be first discussed within asterisk-dev.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835 |
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atis at iq-labs.net
Guest
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| Posted: Mon Jun 02, 2008 9:08 am Post subject: [asterisk-users] Realtime SIP registration for multiple Aste |
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On Mon, Jun 2, 2008 at 4:24 PM, Mindaugas Kezys <mkezys at gmail.com> wrote:
| Quote: | I need this solution and I'm interested in developing such functionality.
What would be the best way to do this, how do you think?
Sip reload is a must. I can initiate it through GUI using AMI.
I'm not familiar with #exec stuff and can't find it described anywhere.
If I understand you correctly - it is possible to include some script in
sip.conf file which can generate "registry =>" lines from DB?
That would be the solution. No additional development would be necessary.
Can somebody point me to the info about #exec stuff?
|
Hi,
I haven't played with it, but as i've heard, you should add #exec
my_script at top of file, and as i just found in google - add
execincludes=yes in asterisk.conf
Then, the script "my_script" should just print corresponding config
lines in stdout, which will be processed by Asterisk.
Regards,
Atis
| Quote: |
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
| Quote: | -----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
bounces at lists.digium.com] On Behalf Of Atis Lezdins
Sent: Monday, June 02, 2008 2:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Realtime SIP registration for multiple
Asterisk servers
On Mon, Jun 2, 2008 at 2:11 PM, Mindaugas Kezys <mkezys at gmail.com>
wrote:
| Quote: | Hello,
We are building solution from multiple Asterisk servers. They are
|
using
| Quote: | Realtime. Servers can be added dynamically over GUI.
Each server can register to different provider/carrier over SIP.
Question would be: how to make it work using pure Realtime - without
|
editing
| Quote: | .conf files?
We can't use same sip.conf in the Realtime database because servers
|
register
| Quote: | to different provider/carriers and for other server-specific options
(externip/etc).
That could be possible if only 1 Aserisk server would be in the
|
system. What
| Quote: | to do with many Asterisk servers?
Is here any workaround to get registration work with different
|
servers using
Hi Mindaugas,
well, unfortunately there is no way how to add "register" entries in
realtime. Main problem behind this is that those entries are read on
asterisk startup and if You would want to add some peer dynamically,
Asterisk would need to get notified about this.
Only solution that could work is generating "register =>" entries from
database by custom script in "#exec" (i suppose You have to enable
execincludes in asterisk.conf). So, upon start or reload of chan_sip
asterisk will execute your script, which would return correct
providers. So, after adding provider dynamically, You would need to
reload chan_sip.
If you would be interested in developing some kind of mechanism to
allow reading "register" entries from realtime, I could assist You in
some way, as this would be useful to me too, but specification of this
should be first discussed within asterisk-dev.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
Atis Lezdins,
VoIP Project Manager / Developer,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835 |
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