hi everyone, do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)? Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch? --- Vedran Zeljeznak
I had problems with mod_opal but mod_h323 works for me. -Steve On 4 February 2011 12:55, Vedran Zeljeznak <vedran.zeljeznak@gmail.com>wrote:
hi everyone,
do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)?
Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch?
--- Vedran Zeljeznak
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Although there's almost no H323 traffic - we're almost entirely SIP - so I can't say how stable it is for a large load. -Steve On 4 February 2011 19:20, Steven Ayre <steveayre@gmail.com> wrote:
I had problems with mod_opal but mod_h323 works for me.
-Steve
On 4 February 2011 12:55, Vedran Zeljeznak <vedran.zeljeznak@gmail.com>wrote:
hi everyone,
do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)?
Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch?
--- Vedran Zeljeznak
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Use Yate we are try to use mod_opal or mod_h323 in production with-out success, Yate has lot of issue but is a good inter-worker between h323 and sip. Freeswitch has a lost of good think but the inter-worker isn't it. Because of then we use Freeswitch for billing propose but Yate for inter-worker. But if you need use FS for inter-worker use mod_h323 is a little more stable. Best Regards. Gustavo Espeche www.easyipcall.com On Fri, 2011-02-04 at 19:20 +0000, Steven Ayre wrote:
Although there's almost no H323 traffic - we're almost entirely SIP - so I can't say how stable it is for a large load.
-Steve
On 4 February 2011 19:20, Steven Ayre <steveayre@gmail.com> wrote: I had problems with mod_opal but mod_h323 works for me.
-Steve
On 4 February 2011 12:55, Vedran Zeljeznak <vedran.zeljeznak@gmail.com> wrote: hi everyone,
do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)?
Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch?
--- Vedran Zeljeznak
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
I used to use this too. It does work pretty well, for the most part. Q931 clearing causes did have issues though, because Yate converts the SIP response code to an internal table ( http://yate.null.ro/pmwiki/index.php?n=Main.CallEndCauses) and then to a Q931 clearing cause for H323. It meant our customers didn't see the real clearing cause from the SIP Reason header (H323 -> SIP calls), which in our case was a big issue since we needed to pass back DCC 34 which Yate wouldn't give until I made a patched version. -Steve On 4 February 2011 22:09, Gustavo Espeche <gustavo.espeche@upper-soft.com>wrote:
Use Yate we are try to use mod_opal or mod_h323 in production with-out success, Yate has lot of issue but is a good inter-worker between h323 and sip. Freeswitch has a lost of good think but the inter-worker isn't it. Because of then we use Freeswitch for billing propose but Yate for inter-worker. But if you need use FS for inter-worker use mod_h323 is a little more stable.
Best Regards.
Gustavo Espeche www.easyipcall.com
On Fri, 2011-02-04 at 19:20 +0000, Steven Ayre wrote:
Although there's almost no H323 traffic - we're almost entirely SIP - so I can't say how stable it is for a large load.
-Steve
On 4 February 2011 19:20, Steven Ayre <steveayre@gmail.com> wrote: I had problems with mod_opal but mod_h323 works for me.
-Steve
On 4 February 2011 12:55, Vedran Zeljeznak <vedran.zeljeznak@gmail.com> wrote: hi everyone,
do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)?
Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch?
--- Vedran Zeljeznak
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Hi, My vote for Yate + FS http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy works without issue. But i really wish to have it in FS, so we save an extra server.
I used to run yate on the same server as fs... Worked fine. Just assign the server 2 ips. Listen on one with fs sip and yate h323, and listen on the 2nd with yate sip. Steve on iPhone On 5 Feb 2011, at 09:06, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
My vote for Yate + FS
http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
works without issue.
But i really wish to have it in FS, so we save an extra server.
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Hmm good, i'll try... On Sat, Feb 5, 2011 at 1:54 PM, Steven Ayre <steveayre@gmail.com> wrote:
I used to run yate on the same server as fs... Worked fine.
Just assign the server 2 ips. Listen on one with fs sip and yate h323, and listen on the 2nd with yate sip.
Steve on iPhone
On 5 Feb 2011, at 09:06, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
My vote for Yate + FS
<http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy> http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
works without issue.
But i really wish to have it in FS, so we save an extra server.
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Different port numbers on 1 ip might work too... Steve on iPhone On 5 Feb 2011, at 21:53, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hmm good, i'll try...
On Sat, Feb 5, 2011 at 1:54 PM, Steven Ayre <steveayre@gmail.com> wrote: I used to run yate on the same server as fs... Worked fine.
Just assign the server 2 ips. Listen on one with fs sip and yate h323, and listen on the 2nd with yate sip.
Steve on iPhone
On 5 Feb 2011, at 09:06, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hi,
My vote for Yate + FS
http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
works without issue.
But i really wish to have it in FS, so we save an extra server.
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yup thats my idea, because for sip 2 sip i use fs only, for sip -> h323, source sip port (yate) could run on 5061 and h323 -> sip won't create any prob. I just wish that both (fs+yate) will work fine in high call pressure. How many cps and CC were you able to handle? i expect 1000 CC on xecon 8 core, 8 gig ram. On Sat, Feb 5, 2011 at 11:05 PM, Steven Ayre <steveayre@gmail.com> wrote:
Different port numbers on 1 ip might work too...
Steve on iPhone
On 5 Feb 2011, at 21:53, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hmm good, i'll try...
On Sat, Feb 5, 2011 at 1:54 PM, Steven Ayre < <steveayre@gmail.com> steveayre@gmail.com> wrote:
I used to run yate on the same server as fs... Worked fine.
Just assign the server 2 ips. Listen on one with fs sip and yate h323, and listen on the 2nd with yate sip.
Steve on iPhone
On 5 Feb 2011, at 09:06, Saeed Ahmed < <saeedahmad1981@gmail.com> saeedahmad1981@gmail.com> wrote:
Hi,
My vote for Yate + FS
<http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy><http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy> http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
works without issue.
But i really wish to have it in FS, so we save an extra server.
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Similar hardware, but tested with no more than about 25CPS (almost all our traffic had already been moved to use SIP). Sure you'll be able to handle more though. -Steve On 5 February 2011 22:22, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
yup thats my idea, because for sip 2 sip i use fs only, for sip -> h323, source sip port (yate) could run on 5061 and h323 -> sip won't create any prob. I just wish that both (fs+yate) will work fine in high call pressure.
How many cps and CC were you able to handle?
i expect 1000 CC on xecon 8 core, 8 gig ram.
On Sat, Feb 5, 2011 at 11:05 PM, Steven Ayre <steveayre@gmail.com> wrote:
Different port numbers on 1 ip might work too...
Steve on iPhone
On 5 Feb 2011, at 21:53, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
Hmm good, i'll try...
On Sat, Feb 5, 2011 at 1:54 PM, Steven Ayre < <steveayre@gmail.com> steveayre@gmail.com> wrote:
I used to run yate on the same server as fs... Worked fine.
Just assign the server 2 ips. Listen on one with fs sip and yate h323, and listen on the 2nd with yate sip.
Steve on iPhone
On 5 Feb 2011, at 09:06, Saeed Ahmed < <saeedahmad1981@gmail.com> saeedahmad1981@gmail.com> wrote:
Hi,
My vote for Yate + FS
<http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy><http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy> http://voip.null.ro/pmwiki/index.php?n=Main.H323ToSIPSignallingProxy
works without issue.
But i really wish to have it in FS, so we save an extra server.
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Vedran, mod_h323 does work fro you... mod_opal had issues with SIP - H323 interworking T. On Fri, Feb 4, 2011 at 1:55 PM, Vedran Zeljeznak <vedran.zeljeznak@gmail.com
wrote:
hi everyone,
do any of you have any suggestions which H.323 implementation (mod_opal or mod_h323) should be used on a production Freeswitch platform (HEAD version)?
Is it better to forward calls from H.323 trunk through Yate (configured as H.323 to SIP proxy) before terminating them on Freeswitch?
--- Vedran Zeljeznak
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (5)
-
Gustavo Espeche
-
Saeed Ahmed
-
Steven Ayre
-
Tihomir Culjaga
-
Vedran Zeljeznak