VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
| View previous topic :: View next topic |
| Author |
Message |
mail.gery at gmail.com
Guest
|
 Posted: Thu Jan 16, 2014 3:37 pm Post subject: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - |
 |
|
Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our company. I'm in the middle of the planning phase and it turned out that our VoIP provider prefers H.323 protocol for handling voice calls (while SIP is also supported as "plan B").
As I never worked with H.323 channels in Asterisk earlier, I'm not sure if it's stable enough to be used in production.
Googling about the subject didn't help much, I could only find some old and probably outdated information which I don't want to rely on.
Can you please confirm if the OOH323 module in Asterisk 11 is stable enough to use for voice calls? No extra functionality is needed, just to be able to create a H.323 trunk towards the provider and make and receive a maximum of 30 simultaneous voice calls through the trunk.
Thanks for your kind response!
Regards,
Gergely Kiss |
|
| Back to top |
|
 |
paul.belanger at polyb...
Guest
|
| Posted: Thu Jan 16, 2014 7:24 pm Post subject: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - |
|
|
On 14-01-16 03:37 PM, Gergely Kiss wrote:
| Quote: | Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our company.
I'm in the middle of the planning phase and it turned out that our VoIP
provider prefers H.323 protocol for handling voice calls (while SIP is also
supported as "plan B").
As I never worked with H.323 channels in Asterisk earlier, I'm not sure if
it's stable enough to be used in production.
Googling about the subject didn't help much, I could only find some old and
probably outdated information which I don't want to rely on.
Can you please confirm if the OOH323 module in Asterisk 11 is stable enough
to use for voice calls? No extra functionality is needed, just to be able
to create a H.323 trunk towards the provider and make and receive a maximum
of 30 simultaneous voice calls through the trunk.
Thanks for your kind response!
|
Save yourself time / energy and insist using SIP. If your ITSP cannot
accommodate your request, thank them and look for another provider.
H323 is Asterisk is basically dead, sure there is a module, sure it
might compile, but you'll be going down the path of zero help.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
asterisk-list at puzzl...
Guest
|
| Posted: Thu Jan 16, 2014 7:28 pm Post subject: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - |
|
|
On 16-01-14 21:37, Gergely Kiss wrote:
| Quote: | Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as "plan B").
|
It's SIP everywhere and anyone who requires you, in 2014, to use H.323
should get a clue. Avoid them or at least demand SIP.
| Quote: | As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.
|
No idea. Maybe someone else with H.323 experience will respond. AFAIK
it's a dead-end.
Regards,
Patrick
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
Dan_Austin at Phoenix.com
Guest
|
| Posted: Thu Jan 16, 2014 7:58 pm Post subject: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - |
|
|
Patrick Lists wrote:
| Quote: | On 16-01-14 21:37, Gergely Kiss wrote:
| Quote: | Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as "plan B").
|
|
| Quote: | It's SIP everywhere and anyone who requires you, in 2014, to use H.323
should get a clue. Avoid them or at least demand SIP
|
Bah. There is nothing wrong with a working H.323 stack. Just assuming
that they will have a working SIP stack because of the date can lead to
heartache.
| Quote: | | Quote: | As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.
|
|
| Quote: | No idea. Maybe someone else with H.323 experience will respond. AFAIK
it's a dead-end.
|
The ooh323 channel has been fairly reliable in our use case, which involve
connecting to a commercial IP PBX with crud SIP support. Only you can tell
if it will work for you however, as sadly many times new core features only
get tested against the SIP channel(s), or worse only implemented there as
well. Our current Asterisk version is 11.5.1
Dan
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
vlad at mikhelson.com
Guest
|
| Posted: Thu Jan 16, 2014 8:40 pm Post subject: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - |
|
|
On 1/16/2014 6:57 PM, Dan Austin wrote:
| Quote: | Patrick Lists wrote:
| Quote: | On 16-01-14 21:37, Gergely Kiss wrote:
| Quote: | Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as "plan B").
|
It's SIP everywhere and anyone who requires you, in 2014, to use H.323
should get a clue. Avoid them or at least demand SIP
|
Bah. There is nothing wrong with a working H.323 stack. Just assuming
that they will have a working SIP stack because of the date can lead to
heartache.
| Quote: | | Quote: | As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.
|
No idea. Maybe someone else with H.323 experience will respond. AFAIK
it's a dead-end.
|
The ooh323 channel has been fairly reliable in our use case, which involve
connecting to a commercial IP PBX with crud SIP support. Only you can tell
if it will work for you however, as sadly many times new core features only
get tested against the SIP channel(s), or worse only implemented there as
well. Our current Asterisk version is 11.5.1
Dan
|
Sorry, have nothing to say of 11.5 but OOH323 works great in 1.8. I use
it as an Avaya IP Office trunk. No problems.
As you observed for yourself when you researched the topic there is not
a lot of help available, and Asterisk team prefers to make everybody
think that SIP is the only viable call setup protocol around. They kind
of not talking a lot about their own IAX any more.
The official H.323 is abandoned. OOH323 is being supported by a very
capable and responsive guy. He does not frequent the user list as he
subscribes to the developer list, so I normally transfer the help
inquiries to him if there is no traction here.
-Vladimir
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
asterisk-list at puzzl...
Guest
|
| Posted: Thu Jan 16, 2014 10:32 pm Post subject: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - |
|
|
On 17-01-14 01:57, Dan Austin wrote:
| Quote: | Patrick Lists wrote:
| Quote: | On 16-01-14 21:37, Gergely Kiss wrote:
| Quote: | Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as "plan B").
|
|
| Quote: | It's SIP everywhere and anyone who requires you, in 2014, to use H.323
should get a clue. Avoid them or at least demand SIP
|
Bah. There is nothing wrong with a working H.323 stack. Just assuming
that they will have a working SIP stack because of the date can lead to
heartache.
|
By itself there is nothing wrong with a working H.323 stack. I just
would not use it  Using H.323 for one provider while any backup or
alternative providers probably use SIP results in needing two stacks in
testing & production. It also requires the admins to gain knowledge of a
legacy protocol. Maybe there are some incumbents or service providers
with legacy H.323 equipment continuing to offer H.323 service. I get
that. But for a business building a VoIP PBX from scratch H.323 does not
make sense from a cost and operations point of view.
| Quote: | | Quote: | | Quote: | As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.
|
|
| Quote: | No idea. Maybe someone else with H.323 experience will respond. AFAIK
it's a dead-end.
|
The ooh323 channel has been fairly reliable in our use case, which involve
connecting to a commercial IP PBX with crud SIP support. Only you can tell
if it will work for you however, as sadly many times new core features only
get tested against the SIP channel(s), or worse only implemented there as
well. Our current Asterisk version is 11.5.1
|
The OP mentioned that his VoIP provider prefers H.323 so it seems to be
about trunking. IMHO "fairly reliable" is not something that is
acceptable for trunking phone service.
H.323 is what Gopher is to HTTP/webservers. When was the last time you
used a Gopher service? Would you today still buy Gopher based service
because the service provider prefers it?
Regards,
Patrick
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
drdialtone at optonlin...
Guest
|
| Posted: Thu Jan 16, 2014 10:36 pm Post subject: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - |
|
|
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Thursday, January 16, 2014 8:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is
it stable?
On 1/16/2014 6:57 PM, Dan Austin wrote:
| Quote: | Patrick Lists wrote:
| Quote: | On 16-01-14 21:37, Gergely Kiss wrote:
| Quote: | Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out
that our VoIP provider prefers H.323 protocol for handling voice
calls (while SIP is also supported as "plan B").
|
It's SIP everywhere and anyone who requires you, in 2014, to use
H.323 should get a clue. Avoid them or at least demand SIP
|
Bah. There is nothing wrong with a working H.323 stack. Just
assuming that they will have a working SIP stack because of the date
can lead to heartache.
| Quote: | | Quote: | As I never worked with H.323 channels in Asterisk earlier, I'm not
sure if it's stable enough to be used in production.
|
No idea. Maybe someone else with H.323 experience will respond. AFAIK
it's a dead-end.
|
The ooh323 channel has been fairly reliable in our use case, which
involve connecting to a commercial IP PBX with crud SIP support. Only
you can tell if it will work for you however, as sadly many times new
core features only get tested against the SIP channel(s), or worse
only implemented there as well. Our current Asterisk version is
11.5.1
Dan
|
Sorry, have nothing to say of 11.5 but OOH323 works great in 1.8. I use it
as an Avaya IP Office trunk. No problems.
As you observed for yourself when you researched the topic there is not a
lot of help available, and Asterisk team prefers to make everybody think
that SIP is the only viable call setup protocol around. They kind of not
talking a lot about their own IAX any more.
The official H.323 is abandoned. OOH323 is being supported by a very
capable and responsive guy. He does not frequent the user list as he
subscribes to the developer list, so I normally transfer the help inquiries
to him if there is no traction here.
-Vladimir
Hey Vladimir, can you share a bit about the ooH323 trunk to IPO
configuration that's stable?
I've tried a few different setups on both sides and wound up using a PRI to
do it.
A call from the IPO to Asterisk (1.6.2 at the time) would crash the Asterisk
box or not work at all.
I'd love to be able to offer this to my IPO and CM customers!
Thank you!
Pat...
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
|
|
mail.gery at gmail.com
Guest
|
| Posted: Fri Jan 17, 2014 3:31 am Post subject: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - |
|
|
Thank you all for your reply!
I think I'm going to give OOH323 a try. In case I see any functional issues or instability, I'll switch to SIP without spending too much time with debugging.
Regards,
Gergely
On 17 January 2014 02:39, Vladimir Mikhelson <vlad@mikhelson.com (vlad@mikhelson.com)> wrote:
| Quote: |
On 1/16/2014 6:57 PM, Dan Austin wrote:
| Quote: | Patrick Lists wrote:
| Quote: | On 16-01-14 21:37, Gergely Kiss wrote:
| Quote: | Dear List,
I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as "plan B").
|
It's SIP everywhere and anyone who requires you, in 2014, to use H.323
should get a clue. Avoid them or at least demand SIP
|
Bah. There is nothing wrong with a working H.323 stack. Just assuming
that they will have a working SIP stack because of the date can lead to
heartache.
| Quote: | | Quote: | As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.
|
No idea. Maybe someone else with H.323 experience will respond. AFAIK
it's a dead-end.
|
The ooh323 channel has been fairly reliable in our use case, which involve
connecting to a commercial IP PBX with crud SIP support. Only you can tell
if it will work for you however, as sadly many times new core features only
get tested against the SIP channel(s), or worse only implemented there as
well. Our current Asterisk version is 11.5.1
Dan
|
Sorry, have nothing to say of 11.5 but OOH323 works great in 1.8. I use
it as an Avaya IP Office trunk. No problems.
As you observed for yourself when you researched the topic there is not
a lot of help available, and Asterisk team prefers to make everybody
think that SIP is the only viable call setup protocol around. They kind
of not talking a lot about their own IAX any more.
The official H.323 is abandoned. OOH323 is being supported by a very
capable and responsive guy. He does not frequent the user list as he
subscribes to the developer list, so I normally transfer the help
inquiries to him if there is no traction here.
-Vladimir
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
|
|
| Back to top |
|
|
|
|
|
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|
Search
