Re: [asterisk-users] h323-sip: one way connection
oh yes, i'm using h323 not openh323
On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote:
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote:
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad
asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it
says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call
from two side. but it is not good for me because 200 is the name of
extension and when i config asterisk systems, i don't know the name of
extensions, therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.com
wrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146
to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and
'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error
and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
try
UserByAlias=yes in general and type=user in user context.
On Fri, Apr 26, 2013 at 9:48 AM, s m sam.gh1...@gmail.com wrote:
oh yes, i'm using h323 not openh323
On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote:
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote:
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad
asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it
says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call
from two side. but it is not good for me because 200 is the name of
extension and when i config asterisk systems, i don't know the name of
extensions, therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in
h323.conf file? i define the address by host=192.168.0.146 but
asterisk
can not find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad
asghar...@gmail.com wrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146
to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and
'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can
call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error
and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it
says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from
two side. but it is not good for me because 200 is the name of extension
and when i config asterisk systems, i don't know the name of extensions,
therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad
asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
nuFone h323 or openh323?
On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote:
flavor? i do not understand what you mean. please explain more.
thanks
On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote:
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it
says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call
from two side. but it is not good for me because 200 is the name of
extension and when i config asterisk systems, i don't know the name of
extensions, therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad
asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and
'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error
and
how i can solve it?
thanks in advance
sam
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To
Re: [asterisk-users] h323-sip: one way connection
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from
two side. but it is not good for me because 200 is the name of extension
and when i config asterisk systems, i don't know the name of extensions,
therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
what flavor of h323 you are using?
On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote:
thanks Asghar,
i do it, but no thing happened:(
asterisk do not identify host line as ip address of the other end
On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar...@gmail.comwrote:
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from
two side. but it is not good for me because 200 is the name of extension
and when i config asterisk systems, i don't know the name of extensions,
therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from two
side. but it is not good for me because 200 is the name of extension and
when i config asterisk systems, i don't know the name of extensions,
therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf file?
i define the address by host=192.168.0.146 but asterisk can not find it?
why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
try type=peer instead of friend.
On Tue, Apr 23, 2013 at 10:04 AM, s m sam.gh1...@gmail.com wrote:
i know what is the exactly problem. i enable debug for h323 and it says:
could not find user by name 200 or address 192.168.0.146
when i change peer-146 to 200 every thing is ok and i can call from
two side. but it is not good for me because 200 is the name of extension
and when i config asterisk systems, i don't know the name of extensions,
therefore i should use addresses not name of extensions.
do you know how i should define address of the other end in h323.conf
file? i define the address by host=192.168.0.146 but asterisk can not
find it? why?
On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote:
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323-sip: one way connection
please post cli output for both calls.
On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote:
hello everybody
i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk
[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw
this is mu extensions.conf file in 145:
[from-trunk]
exten=_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=_2.,1,Dial(H323/to-146/2${EXTEN:1})
i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default.
if i change peer146 to general, every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have general context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam
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Re: [asterisk-users] h323 with NAT
Danny Nicholas wrote: Hi list, I've been beating my head for about 3 days on this one. I have Asterisk 1.4.41 installed using openh323. As long as I'm inside my firewall, everything is hunky-dory. When I move to server on another subnet, I'm still able to connect, but no longer have sound. Any good pointers or suggestions? Thanks Danny Nicholas I had a similar problem once while using ooh323 with Asterisk 1.4.XX. What I did was to use the most recent version of H323plus with Asterisk and got better results with chan_h323. As (AFAIK) OpenH323 was renamed to H323plus, and several improvements has been made to it, you might want to take a look at it. Note: if you are building Asterisk from source, then the source expects a very old version of OpenH323 and PTLib. You can take a look to the tasks performed by these scripts: http://lists.digium.com/pipermail/asterisk-users/2011-January/258119.html to see how to compile Asterisk with the latest version of H323Plus and PTlib. If you need any additional information about the scripts, just let me know. Regards, -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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
Re: [asterisk-users] h323 with NAT
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jose P. Espinal Sent: Wednesday, April 27, 2011 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] h323 with NAT Danny Nicholas wrote: Hi list, I've been beating my head for about 3 days on this one. I have Asterisk 1.4.41 installed using openh323. As long as I'm inside my firewall, everything is hunky-dory. When I move to server on another subnet, I'm still able to connect, but no longer have sound. Any good pointers or suggestions? Thanks Danny Nicholas I had a similar problem once while using ooh323 with Asterisk 1.4.XX. What I did was to use the most recent version of H323plus with Asterisk and got better results with chan_h323. As (AFAIK) OpenH323 was renamed to H323plus, and several improvements has been made to it, you might want to take a look at it. Note: if you are building Asterisk from source, then the source expects a very old version of OpenH323 and PTLib. You can take a look to the tasks performed by these scripts: http://lists.digium.com/pipermail/asterisk-users/2011-January/258119.html to see how to compile Asterisk with the latest version of H323Plus and PTlib. If you need any additional information about the scripts, just let me know. Regards, -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs [Danny Nicholas] Thanks for the information - but this doesn't seem to play well with SUSE. Any ideas? -- _ -- 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
Re: [asterisk-users] h323 with NAT
[Danny Nicholas] Thanks for the information - but this doesn't seem to play well with SUSE. Any ideas? If you are open to the possibility of building from source I think I might have a little white paper based on the scripts (about installing latest version of H323plus on 1.4.X) by today, after I get home (like 7:30 pm, GMT -4); so you can test with native chan_h323. Meanwhile, do you see anything weird (after enabling 'ooh323 debug') on the CLI? Is there a possibility to test with SIP, to see if the audio problem is explicitly H323 related, and not a networking issue? -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- 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
Re: [asterisk-users] h323 with NAT
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jose P. Espinal Sent: Wednesday, April 27, 2011 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] h323 with NAT [Danny Nicholas] Thanks for the information - but this doesn't seem to play well with SUSE. Any ideas? If you are open to the possibility of building from source I think I might have a little white paper based on the scripts (about installing latest version of H323plus on 1.4.X) by today, after I get home (like 7:30 pm, GMT -4); so you can test with native chan_h323. Meanwhile, do you see anything weird (after enabling 'ooh323 debug') on the CLI? Is there a possibility to test with SIP, to see if the audio problem is explicitly H323 related, and not a networking issue? -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs [Danny Nicholas] Works like a champ with SIP - nothing I can see that is weird on CLI output in H323 -- _ -- 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
Re: [asterisk-users] H323 Trunk Problem calling from Asterisk to Avaya PBX
On Wed, Jun 16, 2010 at 4:35 PM, Shina Owolabi shinacaly...@gmail.comwrote: Hi! I've installed Asterisk 1.4.32 with freepbx-2.6.0 in an attempt to provide a conference bridge for an existing Avaya PBX. I have no control over the Avaya system, but I am able to speak with the admin in charge when I need stuff done. I am running all this in a VirtualBox virtual instance, with CentOS 5.4 as the asterisk's host operating system. I configured a h323 trunk asterisk based on a few guides I discovered online, and I created a single sip extension (to test), and I am able to make a call from the Avaya PBX extensions successfully to my asterisk-freepbx virtual machine. The problem is when I try to make calls from Asterisk to Avaya, I get no sound whatsover and the call just keeps trying indefinitely until I end it. (I've used Twinkle and Ekiga softphones). This is what I find in the logs: [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Macro [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:12] ExecIf(SIP/16000-0002, 0|AGI|fixlocalprefix) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: ExecIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:13] Set(SIP/16000-0002, OUTNUM=18151) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:14] Set(SIP/16000-0002, custom=AMP) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:15] ExecIf(SIP/16000-0002, 0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: ExecIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:16] Macro(SIP/16000-0002, dialout-trunk-predial-hook|) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/16000-0002, ) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Macro [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:17] GotoIf(SIP/16000-0002, 0?bypass|1) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:18] GotoIf(SIP/16000-0002, 1?customtrunk) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Goto (macro-dialout-trunk,s,21) [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:21] Set(SIP/16000-0002, pre_num=AMP:h323/Avaya/) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:22] Set(SIP/16000-0002, the_num=OUTNUM) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:23] Set(SIP/16000-0002, post_num=@ 10.100.7.15:1720) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:24] GotoIf(SIP/16000-0002, 1?outnum:skipoutnum) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Goto (macro-dialout-trunk,s,25) [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: GotoIf [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:25] Set(SIP/16000-0002, the_num=18151) in new stack [Jun 16 15:29:27] DEBUG[8721] app_macro.c: Executed application: Set [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Executing [...@macro-dialout-trunk:26] Dial(SIP/16000-0002, h323/Avaya/18...@10.100.7.15:1720|300|) in new stack [Jun 16 15:29:27] VERBOSE[8721] logger.c: -- Requested transfer capability: 0x00 - SPEECH my h323.conf file is below: [general] port = 1720 bindaddr = 10.101.4.224 amaflags = AVAYA progress_setup = 8 progress_alert = 8 faststart = yes h245tunneling = yes gatekeeper = DISABLE disallow=all allow=g729 allow=g723 dtmfmode=rfc2833 context=from-internal h323id=ObjSysAsterisk callerid=testbridge logfile=/var/log/asterisk/h323_log [Avaya] type=friend context=from-internal host=10.100.7.15 port=1720 disallow=all allow=g729 allow=g723 canreinvite=no dtmfmode=rfc2833 Please help me find out why the call isn't going through. -- best regards, Sina Owolabi 2348034022578 23417203257 23417420690 -- best regards, Sina Owolabi 2348034022578 23417203257 23417420690 --
Re: [asterisk-users] H323 RTP Transmission error of packet
Nobody on this ? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi Sent: September-16-09 7:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323 RTP Transmission error of packet Using H323 to reach another h323 switch, I have no audio and the following error: [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21282 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21283 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21284 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21285 to XXX.XXX.XXX.XXX:6064: Invalid argument [Sep 16 15:45:55] DEBUG[10528]: rtp.c:2760 ast_rtp_raw_write: RTP Transmission error of packet 21286 to XXX.XXX.XXX.XXX:6064: Invalid argument Can you please tell me what I`m missing I`m doing a quick dial like Dial(h323/1514...@xxx.xxx.xxx.xxx) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 situation
Hi, Still I can manage to have good incoming calls from h323. Can someone give me a hand? Regards, LS Date: Thu, 16 Jul 2009 15:46:43 +0100 From: Luis Silva luis.si...@dreamware.pt Subject: [asterisk-users] H323 situation To: asterisk-users@lists.digium.com Message-ID: 00ab01ca0624$3c9f69b0$b5de3d...@silva@dreamware.pt Content-Type: text/plain; charset=us-ascii Hi all, I have this installation: Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3 and openh323_v1_18_0. I have a problem that is, when a call comes from H323 and goes to a Sip phone the asterisk sends two rtp streams to the sip. I checked this with tcpdump, save the payload (voice is in G711u), one is the ringing indication and the other is the voice coming from the user in h323 side. And worst they go to the same port. This causes that in the sip phone there are problems, when the call is answered sometimes we get the riging indication, others a mix of the two with very bad sound quality and others(few) a god audio call. The outgoing calls from sip to H323 are ok. I also tested an incoming call from a dahdi channel and from here everything is ok, only one rtp stream and a good call. By the way I had other problem that I fixed, but don't know if it was in the best way. The h323 box is a Cisco AS5300 (or 5350?) and when I was making outgoing calls the AS disconnected all of them after 10 sec. I investigated I noticed that the AS as a limitation to the G711 payload to 20 ms, and asterisk was using 150 ms. I resolve this changing in frame.c the codec value and recompile asterisk. There is simpler way to do this? Like changing values in codec.conf?... Regards LS ___ -- 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
Re: [asterisk-users] h323 guide for asterisk
Maybe this can help you? http://astrecipes.net/index.php?n=286 Thanks l. 2009/5/31 Tamer Higazi th9...@googlemail.com Hi people! I am looking for a h.323 implementation guide for asterisk. I looked in the doc folder of the latest asterisk source distribution and I didn't fund anything acording to this subject. If you guys could give me any advise, I would thank you. Tamer ___ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- 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
Re: [asterisk-users] H323
Yes, this has already been answered. Search previous post for implementation. On Thu, Oct 9, 2008 at 3:34 AM, michel freiha [EMAIL PROTECTED] wrote: Dear all, Does asterisk supports H323?If yes how to enable it? Regards ___ -- 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 ___ -- 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
Re: [asterisk-users] H323 protocol
Yes you can. Obviously you have to compile, configure and add chan_h323 to Asterisk. Map On Thu, Aug 28, 2008 at 10:32 AM, mahboob zaman [EMAIL PROTECTED]wrote: hi. i have two IP phones that are in H323 protocol. How can i test that these two phones are working? For IP phone (SIP) i used asterisk server. can i use asterisk server to test the ip phone with H323 protocol. -- Mahboob Zaman System Engr Systems Services Limited Cell: +8801712280308 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 protocol
Hi, Thanks for reply. can u give me information in detail? How can i compile and can i add chan_h323 ? Thanks mahboob On 8/28/08, map [EMAIL PROTECTED] wrote: Yes you can. Obviously you have to compile, configure and add chan_h323 to Asterisk. Map On Thu, Aug 28, 2008 at 10:32 AM, mahboob zaman [EMAIL PROTECTED] wrote: hi. i have two IP phones that are in H323 protocol. How can i test that these two phones are working? For IP phone (SIP) i used asterisk server. can i use asterisk server to test the ip phone with H323 protocol. -- Mahboob Zaman System Engr Systems Services Limited Cell: +8801712280308 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mahboob Zaman System Engr Systems Services Limited Cell: +8801712280308 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 protocol
http://www.voip-info.org/wiki/view/Asterisk+H323+channels Google is your friend. PC --- Paul Catchpole CCNA Cisco Enterprise Network Consultant Bluecat Certified Engineer www.paulcatchpole.co.uk 0121 285 _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mahboob zaman Sent: 28 August 2008 12:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] H323 protocol Hi, Thanks for reply. can u give me information in detail? How can i compile and can i add chan_h323 ? Thanks mahboob On 8/28/08, map [EMAIL PROTECTED] wrote: Yes you can. Obviously you have to compile, configure and add chan_h323 to Asterisk. Map On Thu, Aug 28, 2008 at 10:32 AM, mahboob zaman [EMAIL PROTECTED] wrote: hi. i have two IP phones that are in H323 protocol. How can i test that these two phones are working? For IP phone (SIP) i used asterisk server. can i use asterisk server to test the ip phone with H323 protocol. -- Mahboob Zaman System Engr Systems Services Limited Cell: +8801712280308 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mahboob Zaman System Engr Systems Services Limited Cell: +8801712280308 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 protocol
El jue, 28-08-2008 a las 01:32 -0700, mahboob zaman escribió: hi. i have two IP phones that are in H323 protocol. How can i test that these two phones are working? For IP phone (SIP) i used asterisk server. can i use asterisk server to test the ip phone with H323 protocol. I've wrote a small guide to enable chan_h323.so on asterisk 1.4 (is in spanish, sorry): http://www.ecualug.org/?q=2008/04/18/comos/asterisk_14_agregando_soporte_h323_chan_h323so_en_asterisk_14 Best regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net SIP : [EMAIL PROTECTED] FWD : 558563 USA : 1 360 968 1701 Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 Issue
On Sunday 10 August 2008 13:31:22 emist wrote: I'm not sure whats going on but I have built and installed chan_ooh323 from asterisk addons. When I try to dial a call to an h323 provider i get the Channel not implemented error. When I load chan_ooh323.so I get: [Aug 10 14:28:00] WARNING[23007]: loader.c:647 load_resource: Module 'chan_ooh323.so' already exists. Which seems to indicate its already installed and loaded. However, when I check what channel types are available h323 doesn't appear: If the config file does not exist or if it contains insufficient data, then the channel type will not register. Try running 'module reload chan_ooh323.so', and fix any errors displayed as a result. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 Issue
Thanks Tilghman, that was the issue. Regards, Igor H. Tilghman Lesher wrote: On Sunday 10 August 2008 13:31:22 emist wrote: I'm not sure whats going on but I have built and installed chan_ooh323 from asterisk addons. When I try to dial a call to an h323 provider i get the Channel not implemented error. When I load chan_ooh323.so I get: [Aug 10 14:28:00] WARNING[23007]: loader.c:647 load_resource: Module 'chan_ooh323.so' already exists. Which seems to indicate its already installed and loaded. However, when I check what channel types are available h323 doesn't appear: If the config file does not exist or if it contains insufficient data, then the channel type will not register. Try running 'module reload chan_ooh323.so', and fix any errors displayed as a result. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h323 channel compile error
Hi, I have following settings done on my Fedora8: Downloaded openh323-v1_19_0_1-src-tar.gz pwlib-v1_11_1-src.tar.gz to my knowledfe chan_h323 should be compiled against openh323-v1_18_0-src.tar.gz and pwlib-v1_10_3-src-tar.gz cheers -- Daniele Santi .o. [EMAIL PROTECTED] ..o () ascii ribbon campaign Linux User #415108 ooo /\ www.asciiribbon.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 registeration and routing the calls
I have not tested it but in theory you should be able to authorize it by setting host= in the peer details. - Original Message - From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, November 09, 2007 11:14 PM Subject: [asterisk-users] H323 registeration and routing the calls Hi All; As I understood that h323 module in asterisk does not support the ability to let the h323 endpoints register at asterisk (this registeration happens at 1719 port), so how asterisk will be able to route the call for the destination IP Phone if it is not registered (so the IP is unknown)? I do not know if current h323 module supports registeration via 1719 port. Any help? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 ___ --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
Re: [asterisk-users] h323 problem with asterisk 1.2.18
Instead of using those H323. chan drivers try using the ones in asterisk-addons-1.2.16. They seemed to work a lot better for me than the ones that came with the main asterisk package. - Original Message - From: nik600 [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, May 07, 2007 8:40 PM Subject: [asterisk-users] h323 problem with asterisk 1.2.18 i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/ export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH but when i go to: cd asterisk-1.2.18/channels/h323/ and do a make opt: [EMAIL PROTECTED]:/data/programmi/asterisk_1.2.18/asterisk-1.2.18/channels/h323# make opt make: *** No rule to make target `opt'. Stop. why? where am i wrong? i've also tried the last version of pwlib and openh323, but without fixing the problem thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h323
Did you compile H.323 for asterisk and then make install asterisk ? - Original Message - From: Pezhman Lali [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, March 28, 2007 4:30 PM Subject: [asterisk-users] h323 hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI -- Executing Dial(SIP/2.2.2.2-086f5ac0, H323/[EMAIL PROTECTED]|60) in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28 14:17:23 NOTICE[11985]: app_dial.c:1059 dial_exec_full: Unable to create channel of type 'H323' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/2.2.2.2-086f5ac0' status is 'CHANUNAVAIL' Don't get soaked. Take a quick peek at the forecast with the Yahoo! Search weather shortcut. http://tools.search.yahoo.com/shortcuts/#loc_weather ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h323 how to set it up?
Florea Igor wrote: Hi all, I have some questions about h323. Is it mandatory to install a oh323 or I can do h323 calls without patching or adding any new drivers ti asterisk? I did compile the asterisk with channel driver chan_h323 but it still gives me this error: [Feb 28 18:12:58] WARNING[1902]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'H323' (cause 66 - Channel not implemented) what shoul I do to have it implemented? Can somebody recommend some references on how to set up h323 ? Thx, Igor This message was scanned by Barracuda Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Read README file in channels/h323 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] H323-to-SIP proxy
T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323-to-SIP proxy
What about the SIP leg? - Mensaje Original - De: Michelle Dupuis [EMAIL PROTECTED] Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviados: martes 27 de febrero de 2007 16h'08 (GMT-0300) America/Argentina/Buenos_Aires Asunto: RE: [asterisk-users] H323-to-SIP proxy T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] H323-to-SIP proxy
T.38 pass-through should work fine on the SIP leg. (With Asterisk 1.40) There are a few bugs but you can get past them. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 2:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] H323-to-SIP proxy What about the SIP leg? - Mensaje Original - De: Michelle Dupuis [EMAIL PROTECTED] Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviados: martes 27 de febrero de 2007 16h'08 (GMT-0300) America/Argentina/Buenos_Aires Asunto: RE: [asterisk-users] H323-to-SIP proxy T.38 won't work over the H.323 leg of your call (even with Open H.323), chan_h323 won't support it. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Octavarium Sent: Tuesday, February 27, 2007 12:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] H323-to-SIP proxy I need to receive a FAX call from a SIP device into my Asterisk box, then send that FAX call to an H323 gateway and bridge the call, so Asterisk will be acting as a Converter. SIP device is a Grandstream HT496 so i can configure FAX Pass-through, but the H323 gateway only supports T.38 BTW, i am able to make voice calls from SIP device to H323 gateway, the problem is with FAX How can i do this? Best Regards, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 to SIP - One way voice
Which H.323 channel driver are you using, and could you post a log or debug of a session. Craig - Original Message - From: Andrei U [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, February 08, 2007 2:41 AM Subject: [asterisk-users] H323 to SIP - One way voice Hello all, I want to use asterisk as protocol converter, H323 to SIP. I am using Asterisk 1.2.14 with chan_h323 and the free version of g729. When calling from SIP to H323 everything is fine. But when calling from H323 to SIP, the phone using SIP doesn't hear the other party. The phones and Asterisk are in the same subnet and the firewall of the Asterisk box is off. Please advice. Thank you, Andrei U ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] h323 compile error
I thinik the code is too new for your compiler... I remember reading about needing GCC 2.95 somewhere... I'm just about to post on a similar theme! I am getting the following compile error on h323. Its an old redhat 7.3 system with asterisk 1.2.14, zaptel 1.2.12 pwlib 1.5.2 and openh323 1.12.2 I have pwlib compiled and installed. I have openh323 compiled and installed. I went in the channels/h323 directory and did make opt What shall I do? Jerry ../../include/asterisk/utils.h: In function `void ast_slinear_saturated_divide (short int *, short int *)': ../../include/asterisk/utils.h:199: warning: `always_inline' attribute directive ignored ../../include/asterisk/utils.h: In function `int inaddrcmp (const sockaddr_in *, const sockaddr_in *)': ../../include/asterisk/utils.h:217: warning: `always_inline' attribute directive ignored In file included from ast_h323.cxx:51: ast_h323.h: At top level: ast_h323.h:159: type specifier omitted for parameter ast_h323.h:159: parse error before `*' ast_h323.cxx:957: type specifier omitted for parameter ast_h323.cxx:957: parse error before `*' ast_h323.cxx: In method `H323Channel *MyH323Connection::CreateRealTimeLogicalChannel (...)': ast_h323.cxx:959: `capability' undeclared (first use this function) ast_h323.cxx:959: (Each undeclared identifier is reported only once for each function it appears in.) ast_h323.cxx:959: `dir' undeclared (first use this function) ast_h323.cxx:959: `sessionID' undeclared (first use this function) make: *** [ast_h323.o] Error 1 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323 NAT Problem
I dont think the registration will be the problem, but the media communication, for that you could use an Application Layer Gateway (ALG), you can check netfilter.org for more information. Regards On 12/1/06, Jason Kim [EMAIL PROTECTED] wrote: Hi, I installed asterisk with oh323. My gatekeeper is out of nat device. How can i register * to gatekeeper? Thanks in advance.. Jason. Cheap talk? Check out Yahoo! Messenger's low PC-to-Phone call rates. http://voice.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323
On Sunday 27 August 2006 10:40, Mohammad Salaque wrote: any one try that with g723 codec? We use G.723.1, and it works well. My only problem is the bridging time (after pickup) takes at least 5 seconds. But this happenned even before Asterisk was in the picture, so I'm guessing it's the remote H.323 gateways (unless someone else has experienced this). Cheers, Mark. pgpA2t5GdGVZR.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323
any one try that with g723 codec? thanks Salaque On 8/27/06, Rosli Sukri [EMAIL PROTECTED] wrote: i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeeting On 8/26/06, atik khan [EMAIL PROTECTED] wrote: Hi, i used to work ooh323 with my asterisk. it gives better performance than other oh323 or H323 comes with asterisk... i got H323 channel and oh323 with a lot of error.( like codec selection )but ooh323 works fine with me thanks atik On 26 Aug 2006 12:13:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Got my Gmail ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323
Hi, i used to work ooh323 with my asterisk. it gives better performance than other oh323 or H323 comes with asterisk... i got H323 channel and oh323 with a lot of error.( like codec selection )but ooh323 works fine with me thanks atik On 26 Aug 2006 12:13:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] H323
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeetingOn 8/26/06, atik khan [EMAIL PROTECTED] wrote:Hi,i used to work ooh323 with my asterisk. it gives better performance than otheroh323 or H323 comes with asterisk...i got H323 channel and oh323 with a lot of error.( like codecselection )but ooh323 works fine with methanksatikOn 26 Aug 2006 12:13:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 Asterisk best practices
Joshua Laroff wrote: I recently have been required to terminate traffic via H323. We have beensuccessfully handling this traffic as SIP. We often have 30 + concurrent calls on this server and I am not quite sure the best way to handle this new H322 traffic. Which of the h323 channels for * can handle this traffic reliably? Any suggestions would be greatly appreciated. Thanks, JC -- Hi JC, oh323, which uses OpenH323 is pretty solid and reliable from inaccessnetworks. I like it much more than the other two. There is also something called chan_woomera, a new channel for Asterisk which can hook up to OpenH323 or Opal. try it! -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
Finally I installed the oh323 without any errors and tested that with SJPhone.(Played the demo message). Now my question is, it seems from any h323 client anyone can make calls to my asterisk if they dial number@my serverip. How do I do the authentication by IP, username, password like SIP.conf and IAX.conf? Any help would be appreciated. Thanks, ThameemOn 6/8/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jun 08, 2006 at 08:23:15PM -0700, Thameem Ansari wrote: Hello guys, Thanks for your replies. I finally got the ooh323 built successfully. But again the problem is I am using sjphone to connect to my server. I can initiate the call which rings the phone without any problem. But its keep on ringing even if I take the call. I dunno whats goin on? Simply this h323 configuration suckssjphone is a SIP phone, right? Why don't you start with calling an echo test extension from the h323phone? Or generate such a call from the server (using a .call file orOriginate in the manager).--Tzafrir Cohen sip:[EMAIL PROTECTED]icq#16849755 iax:[EMAIL PROTECTED]+972-50-7952406[EMAIL PROTECTED] http://www.xorcom.com___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
Hello all, I am trying to use native h323 built from asterisk 1.2.7. I configured the h323 to receive incoming calls...the problem is i can receive the call to my asterisk and it rings another extension but no audio. I don't see any good documentation about gatekeepers, fast start, etc with h323. I would like to get some help from you guys to fix this issue. If any of you have configured asterisk with h323, please help me do that. Thanks in advance, Thameem Hi Thameem, I had a similiar problem, so try different combinations of faststart, h245Tunnelling,h245inSetup. Also, you can try the ooh323 in asterisk-addons, or oh323 from in-access networks. thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
Hi yousuf, Please tell me to make h323 work, what are the other things i need to do other than getting the chan_h323.so under modules? Do I need to install OpenGatekeeper and configure it ? Do I need fast start? fast tunneling? h245inSetup? (I really don't have any idea about what these components are) Thanks, ThameemOn 6/8/06, Yusuf [EMAIL PROTECTED] wrote: Hello all, I am trying to use native h323 built from asterisk 1.2.7. I configured the h323 to receive incoming calls...the problem is i can receive the call to my asterisk and it rings another extension but no audio. I don't see any good documentation about gatekeepers, fast start, etc with h323. I would like to get some help from you guys to fix this issue. If any of you have configured asterisk with h323, please help me do that. Thanks in advance, ThameemHi Thameem,I had a similiar problem, so try different combinations of faststart,h245Tunnelling,h245inSetup.Also, you can try the ooh323 in asterisk-addons, or oh323 from in-access networks.thanks,yusuf--This message has been scanned for viruses anddangerous content by MailScanner, and isbelieved to be clean.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
If I were you, I would install the lastest asterisk-addons. there is an asterisk ooh323c directory , read the REAME on that directoryThameem Ansari [EMAIL PROTECTED] wrote: Hello All, Somereason my previous mail was not get into the list (or may be delayed). I have a problem successfully configuring the h323 support with asterisk 1.2.7. I searched the net and I don't find any useful or clear documentation. First tell me, which h323 installation should I go with? h323 (native) or open h323 or OOH323? Secondly, How do I configure h323 (any version) with already running asterisk? If I could get some success stories that would shed some light on my efforts. Thanks in advance, Thameem ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
It seems that Open H323 only work with Asterisk version 1.0. As per the latest stable README of asterisk-oh323 here is the readme. Required packages --- In order to build the OH323 Asterisk channel driver you will need some other packages. We recommend to download their source and build them. These are the following: o PWlib (Portable Text and GUI C/C++ Class Library) download from http://sourceforge.net/projects/openh323 (v1.8.7/Mimas_patch2) (required) o OpenH323 (Class Library implementing the H.323 protocol) download from http://sourceforge.net/projects/openh323 (v1.15.6/Mimas_patch2) (required) o Asterisk PBX (Open Source Linux PBX) download from http://www.asterisk.org (CVS v1-0, 2005-09-08) (required) o OhPhone (Command line H.323 client) download from http://www.openh323.org (v1.13.5) (optional, used for testing) o OpenH323 Gatekeeper (H.323 Gatekeeper) download from http://www.gnugk.org (v2.2.2) (optional, used for testing) Although the usage of a gatekeeper is optional, it is recommended for easier address translation. This software has been developed and tested with the aforementioned versions of the above packages. Using other versions may break things, so try these versions first. Anybody has any idea I want to compile this with asterisk 1.2.7 and 1.2.8 Thanks, Thameem On 6/8/06, Thameem Ansari [EMAIL PROTECTED] wrote: Hi yousuf, Please tell me to make h323 work, what are the other things i need to do other than getting the chan_h323.so under modules? Do I need to install OpenGatekeeper and configure it ? Do I need fast start? fast tunneling? h245inSetup? (I really don't have any idea about what these components are) Thanks, ThameemOn 6/8/06, Yusuf [EMAIL PROTECTED] wrote: Hello all, I am trying to use native h323 built from asterisk 1.2.7. I configured the h323 to receive incoming calls...the problem is i can receive the call to my asterisk and it rings another extension but no audio. I don't see any good documentation about gatekeepers, fast start, etc with h323. I would like to get some help from you guys to fix this issue. If any of you have configured asterisk with h323, please help me do that. Thanks in advance, ThameemHi Thameem,I had a similiar problem, so try different combinations of faststart,h245Tunnelling,h245inSetup.Also, you can try the ooh323 in asterisk-addons, or oh323 from in-access networks.thanks,yusuf--This message has been scanned for viruses anddangerous content by MailScanner, and isbelieved to be clean.___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
On Thu, Jun 08, 2006 at 04:58:20PM -0700, Thameem Ansari wrote: It seems that Open H323 only work with Asterisk version 1.0. As per the latest stable README of asterisk-oh323 here is the readme. Which h323? chan_oh323 is just one of at least three h323 channels. Versions 0.7x of it are for Asterisk 1.2 , and is distributed independently of Asterisk. The directory asterisk/channels/h323 includes chan_h323 . And addons package includes chan_ooh323c . Unlike the latter two it does not use openh323 and thus a lot simpler to build (assuming you have gcc-objc). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
And addons package includes chan_ooh323c . Unlike the latter two it does not use openh323 and thus a lot simpler to build (assuming you have gcc-objc). gcc-objc? IIRC, the Objective System OOH323 is written in plain C(99?) not Objective-C. If they wrote it in Objective-C, they would be obliged to name OOH323X :). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
Hello guys, Thanks for your replies. I finally got the ooh323 built successfully. But again the problem is I am using sjphone to connect to my server. I can initiate the call which rings the phone without any problem. But its keep on ringing even if I take the call. I dunno whats goin on? Simply this h323 configuration sucks -ThameemOn 6/8/06, Leo Ann Boon [EMAIL PROTECTED] wrote: And addons package includes chan_ooh323c . Unlike the latter two it doesnot use openh323 and thus a lot simpler to build (assuming you havegcc-objc).gcc-objc? IIRC, the Objective System OOH323 is written in plain C(99?) not Objective-C. If they wrote it in Objective-C, they would be obligedto name OOH323X :).___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 with asterisk problem
On Thu, Jun 08, 2006 at 08:23:15PM -0700, Thameem Ansari wrote: Hello guys, Thanks for your replies. I finally got the ooh323 built successfully. But again the problem is I am using sjphone to connect to my server. I can initiate the call which rings the phone without any problem. But its keep on ringing even if I take the call. I dunno whats goin on? Simply this h323 configuration sucks sjphone is a SIP phone, right? Why don't you start with calling an echo test extension from the h323 phone? Or generate such a call from the server (using a .call file or Originate in the manager). -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 to sip ringing indication
On Saturday 20 May 2006 16:31, Roman Yeryomin wrote:: Hello all! I have a problem with ringing indication when calling from h323 (oh323+open phone client) to sip users. The phone rings and users can talk to each other with no problems but the calling h323 user hear silence unless sip user picks up the phone. Calling to pstn no problems. Calling from sip to that open phone client also no problems. I tried latest ooh323 and oh323... no difference Also passing r option to dial doesn't help. Does anyone know where could be the problem? Roman That's strange, but it's working now... I didn't change anything.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 calls will not stay connected
Daren J. Howell DTCommunication wrote: I have restricted the asterisk server to G711 to match the choice on the PBX, and still same result. I have read that either endpoint have to be either a master or slave to communicate to each other. I see in the logs that both are shown to be a slave. The pbx side has to be set to slave. How can I lock the asterisk side to be a master? Or is this something to worry about? Hi Daren, I believe the endpoints negotiate the master slave thing, so I'm not sure this is the issue here. I had the exact same problem when I set up and it was caused by a codec mismatch, but I'm sure there are other factors that will give the same result. Sorry I can't offer any more. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 calls will not stay connected
Daren J. Howell DTCommunication wrote: Have Asterisk connected to a H323 compatible legacy PBX using QSIG protocol and IP trunks. I can call to Asterisk, and from Asterisk using X-Lite softphone but whenever either end picks up, the calls disconnects. Try restricting both ends to one codec; disallow=all allow=codec of choice at the asterisk end and whatever you need to do at the legacy end. Regards, Richard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 to SIP
Farhad Ibragimov wrote: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me Asterisk is perfectly documented everywhere on the net. Maybe the first place to visit in order to have working asterisk is www.asterisk.org.Second place is www.voip-info.org If any question arises feel free to email me privately. Tofik Suleymanov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 to SIP
You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 to SIP
I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 to SIP
You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 to SIP
Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 to SIP
On Sun, 7 May 2006 19:58:26 +0500, Farhad Ibragimov [EMAIL PROTECTED] wrote: Thanks Try reading this URL (spanish language): http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323 With the page instructions I can call from and to H.323 to every registred SIP/IAX2/H.323 device with my Asterisk box. Good luck, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could begin with: http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation http://www.voip-info.org/wiki/view/Asterisk+H323+channels http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels and much more. You need to install chan_h323 module and configure as well as you need in your application, (if you need gatekeeper functionality maybe you need to try before GNUGK), and later via extensions make wherever you need. Its a little complicated and you need how to work with asterisk before doing all this things. Regards Farhad Ibragimov escribió: I don’t have practice to work with Asterisk but I see that is a great soft. If you have any idea or some config files can you help me -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Sagredo Sent: Sunday, May 07, 2006 7:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H323 to SIP You could make a H323 to SIP transport. Before to do that, you need to have installed and working both chan protocolos on Asterisk. aFarhad Ibragimov escribió: Hi all I have installed station which support only H323 protocol. I want to install SIP telephone. Is it possible to call SIP telephone throught my station ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo V. Salas M Telconet S.A. Calle 15 y Avenida 24 Esquina Edificio Barre #2 1er Piso Teléfono: 262 8071 Celular : 09 985 5138 Manta - Manabà - Ecuador ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
This is my debugtried with netmeeting I can still hear but when I talk nothing happenmygw*CLI h.323 debugH323 debug enabledmygw== New H.323 Connection created. -- Setting up Call -- oCall token: [ip$192.168.1.219:1057/8226] -- oCalling party name: [myPersonal] -- oCalling party number: [] -- oCalled party name: [ip$192.168.1.214:1720] -- oCalled party number: []mygw--Received SETUP messageAllowed Codecs: mygw Table: G.729A 1 G.729 2 G.723.1 3 G.711-uLaw-64k 4 G.711-ALaw-64k 5 UserInput/hookflash 6 UserInput/RFC2833 7 Set:aco*CLI 0:aco*CLI 0:o*CLI G.729A 1 G.729 2 G.723.1 3 G.711-uLaw-64k 4 G.711-ALaw-64k 5 1:o*CLI UserInput/hookflash 6 2:o*CLI UserInput/RFC2833 7mygw*CLImygw=-= In OnAnswerCall for call 8226mygw*CLI - Progress Indicator: 0mygw*CLI - Inserting PI of 0 into ALERTING message == Starting H323/ip$192.168.1.219:1057/8226 at default,ip$192.168.1.214:1720,1 failed so falling back to exten 's' -- Executing Playback(H323/ip$192.168.1.219:1057/8226, demo-echotest) in new stack mygwAnswering call ip$192.168.1.219:1057/8226 -- Playing 'demo-echotest' (language 'en')mygw-- Started logical channel: sending G.711-uLaw-64kmygw*CLI -- channelsOpen = 1mygw=-= In OnConnectionEstablished for call 8226 mygw*CLI -- Connection Established with myPersonal [192.168.1.219]mygwMyH323_ExternalRTPChannel::OnReceivedAckPDUmygw*CLI -- remoteIpAddress: 192.168.1.219 -- remotePort: 49600 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 192.168.1.219 -- remotePort: 49600 -- ExternalIpAddress: 192.168.1.214 -- ExternalPort: 17950mygw*CLI . -- Executing Echo(H323/ip$192.168.1.219:1057/8226, ) in new stack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
I am not sure if this debug message is enough information. Try to do what I told. Switch to another H323 channel driver and see what happens. Try first chan_oh323. Michael Mansos(or something like that) and other guys have been done a good job. Isamar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
Hello, As I know from my experience with Chan323 and OH323. I setup Asterisk on Redhat 9.0 i386 and it is working without any problem with Chan323, OH323 libraries required. I never tried OOH323 come (0.4) with Asterisk. If possible I would like to know how to use newest version of OOH323 (0.8.1) with Asterisk? boldsoft*CLI show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by [EMAIL PROTECTED] on a i686 running Linux Interesting thing is even sometimes X-Lite doesn't work properly sometimes if both end behind NAT. But If I to use X-Pro one-end and other-end X-Lite, this case working normal. But Audiocodes, Addpac, Davolink and other gateways with G729, G723 working without any problem with Chan323 and OH323. I did upgrade Asterisk from existing version to latest 1.2.6 and installed Chanh323 and OH323 0.7.3 with neccessary libraries. Both work one-way voice only, when I to use X-Lite and X-Pro. I don't know how to get work Chan323 and OH323 with Asterisk 1.2.6. Regards, Balgaa - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, April 02, 2006 8:08 PM Subject: Re: [Asterisk-Users] H323 on way voice I am not sure if this debug message is enough information. Try to do what I told. Switch to another H323 channel driver and see what happens. Try first chan_oh323. Michael Mansos(or something like that) and other guys have been done a good job. Isamar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
Is the SIP phone behind NAT? That's one of the common reasons for one way audio. You might want to try forwarding some port ranges if you are behind NAT just to eliminate that as a possiblity. The SIP port ranges should be something like: SIP: 5060-5061RTP: 1-2KyleOn 4/1/06, Il Neofita [EMAIL PROTECTED] wrote: Hi,I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
Neofita wrote: -- channelsOpen = 1 There is only ONE channel open. This should be a huge alarm to you. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
[EMAIL PROTECTED] wrote: I am not sure if this debug message is enough information. Try to do what I told. Switch to another H323 channel driver and see what happens. Try first chan_oh323. So instead of solving his configuration problem he should try a new channel driver? Michael Mansos(or something like that) and other guys have been done a good job. And myself and the others that have contributed to chan_h323 work haven't? Get a life - Think before you type. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
Good luck. Try to switch between channel drivers. Chan_oh323, chan_h323 and ooh323. and remember to install the *exact* lib versions recommended on the readmes May the force be with you... Isamar On Sat, 1 Apr 2006, Il Neofita wrote: Hi, I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
Il Neofita wrote: Hi, I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? Set a valid bindaddr Ensure G.729 is actually getting allowed If you expect any more assistance, at all, debug information is required - So for now I am totally guessing. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on way voice
On Sat, 1 Apr 2006 20:09:35 -0500, Il Neofita [EMAIL PROTECTED] wrote: Hi, I installed H323, however when I make a call from SIP Phone - Asterisk H323 - Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Try using G.711 oodec. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 behind a Firewall
If you open h323 port and rtp ports, it should work. Il Neofita escribió: There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 behind a Firewall
The h323 channels doesn't have any support for NAT. You'd need to register with a properly configured gnugk for that. Julian J. M. On 3/29/06, Alberto Sagredo [EMAIL PROTECTED] wrote: If you open h323 port and rtp ports, it should work. Il Neofita escribió: There is a proble to put an H323 Asterisk server behind an iptables firewall? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 compilation Help needed
You need to run make from /usr/src/asterisk and not /usr/src/asterisk/channels/h323. Just make then make install. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hussain Umair Sent: Wednesday, January 04, 2006 5:35 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] H323 compilation Help needed hi all im trying to compile h323 i have got the pwlib and openh323 working that is simph323 is running properly but when i try to compile h323 in the channels directory it gives me the following error can anybody please help me with [EMAIL PROTECTED] src]# cd /usr/src/asterisk/channels/h323/ [EMAIL PROTECTED] h323]# make opt g++ -DNDEBUG -I../../include -Wmissing-prototypes -fPIC -DP_LINUX=2.6.5-1.358 -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -DP_USE_PRAGMA -DPHAS_TEMPLATES -I/root/pwlib/include/ptlib/unix -I/usr/include/pwlib -I/root/pwlib/include -DPTRACING -I/root/openh323/include -DHAS_IXJ -DHAS_OSS -fPIC -DP_USE_PRAGMA -Os -DNDEBUG -pipe -x c++ -c ast_h323.cxx -o ast_h323.o ast_h323.cxx:1:1: warning: _GNU_SOURCE redefined command line:4:1: warning: this is the location of the previous definition In file included from ast_h323.cxx:51: ast_h323.h:159: error: type specifier omitted for parameter `RTP_QOS' ast_h323.h:159: error: syntax error before `*' token ast_h323.cxx:957: error: type specifier omitted for parameter `RTP_QOS' ast_h323.cxx:957: error: syntax error before `*' token ast_h323.cxx: In member function `H323Channel* MyH323Connection::CreateRealTimeLogicalChannel(...)': ast_h323.cxx:959: error: `capability' undeclared (first use this function) ast_h323.cxx:959: error: (Each undeclared identifier is reported only once for each function it appears in.) ast_h323.cxx:959: error: `dir' undeclared (first use this function) ast_h323.cxx:959: error: `sessionID' undeclared (first use this function) make: *** [ast_h323.o] Error 1 Thanks alot in advance... _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.13/221 - Release Date: 1/4/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 compilation Help needed
Hi friend, You first need to have the correct verisons of pwlib and openh323 as mentioned in the readme file in ./channels/h323 directory. Note, they have to be the same versions, neither advanced nor otherwise, or else it wont compile. Then you ha ve to give a make from the channels/h323 directory and then give a make install from the asterisk base directory, i.e. /usr/src/asterisk or any other directory where you have untarred asterisk. I am sure of this because I have done it many times during our testing phases. But I recomend you that if you are using h323 protocol, then better use ooh323 which you can download from the internet. Use that package instead of h323 in asterisk. You will need different versions of pwlib and openh323 for it, but it works better than h323 in asterisk and oh323 in asterisk-addons. But remember, if at all you are using oh323 or ooh323, rename the conflicting verisons of chan_h323.so, or chan_oh323.so, or chan_ooh323.so from asterisk modules to something else. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
Prely subjective, but I first installed h323 and it worked. Somewhere along the line something happened and it no longer worked. Recompiling it etc seemed to have no effect. I then tried oh323 and it worked first time and has stayed working. I probably did soemthing wrong, but oh323 seems to work for me. Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Innocent Evil Sent: 05 December 2005 14:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] h323 vs oh323 Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 vs oh323
Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
I am still having a non-solved problem with Oh323/h323 and checking Digium homepage after a long time, it looks like they need some dimes now to support me in this case. I have 46(2 T1) PSTN channels receiving calls through H323 protocol. With oh323, after 40 channels in use, It crashes due to some bug related to the limit of file handles. Even playing with some high values in /proc/sys/fs/file-max, didn't solve. With chan_h323, I don't have this problem but, I have this one: localhost*CLI show channels Channel Location State Application(Data) Zap/20-1 [EMAIL PROTECTED]:1 Up Bridged Call(H323/ip$a.b.c.d) 1 active channel 5 active calls I have only one active channel but 5 active calls?! Asterisk version 1.2.0 with H323 and the same pwlib/H323 libs recommended by the README. Checking the logs, I have tons of these errors: Dec 6 00:36:17 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:18 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:19 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:20 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:21 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! Dec 6 00:36:22 WARNING[31517] channel.c: Avoided deadlock for '0x9cd1380', 10 retries! And this one too: Dec 6 00:36:18 WARNING[31530] channel.c: Prodding channel 'H323/ip$202.83.196.25:32791/31907' failed How to solve this problem? Isamar On Mon, 5 Dec 2005, David Waugh wrote: Prely subjective, but I first installed h323 and it worked. Somewhere along the line something happened and it no longer worked. Recompiling it etc seemed to have no effect. I then tried oh323 and it worked first time and has stayed working. I probably did soemthing wrong, but oh323 seems to work for me. Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Innocent Evil Sent: 05 December 2005 14:36 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] h323 vs oh323 Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works right out of box and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
So, we have h323, oh323 and ooh323 I knew about h323 and oh323 but didn't know about ooh323. What is URL of ooh323, I want to know more about them. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 6 Dec 2005 09:16:05 +1100 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] h323 vs oh323 I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works right out of box and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
Ok.. how many channels are you using? More than 100? Maybe it can be good only for 10 or 20 simultaneous connections... Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works right out of box and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
make http://www.voip-info.org your friend.. http://www.voip-info.org/wiki-Asterisk+H323+channels Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: So, we have h323, oh323 and ooh323 I knew about h323 and oh323 but didn't know about ooh323. What is URL of ooh323, I want to know more about them. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Tue, 6 Dec 2005 09:16:05 +1100 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] h323 vs oh323 I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something else. OOH323 works right out of box and since we started using it to interconnect Asterisk to Samsung OfficeServ 500 we had no problems whatsoever. regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 08:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] h323 vs oh323 Try chan_oh323 and if it is not ok, try chan_h323 Both work well in different situations/equipments. Isamar On Mon, 5 Dec 2005, Innocent Evil wrote: Hello, Would you please share your experience regarding h323 and oh323 in asterisk. I am confused to choose one. Thanks, -- You don't have any choice, you already made it before you came here.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
No, max we used is 30 channels. But according to voip-info its faster protocol because it offloads media processing to asterisk (which is a better choice I think) and only looks after H323 call setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 11:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] h323 vs oh323 Ok.. how many channels are you using? More than 100? Maybe it can be good only for 10 or 20 simultaneous connections... Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323 vs oh323
Ok. I will give one more shot on that. Last time I had one-way-audio issue with that. Thanks. Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: No, max we used is 30 channels. But according to voip-info its faster protocol because it offloads media processing to asterisk (which is a better choice I think) and only looks after H323 call setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 6 December 2005 11:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] h323 vs oh323 Ok.. how many channels are you using? More than 100? Maybe it can be good only for 10 or 20 simultaneous connections... Isamar On Tue, 6 Dec 2005, Boris Bakchiev wrote: I like the chan_ooh323. I like the idea of selfcontained H323 channel that doesn't rely external libraries, often with specific versions that conflict with something ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 question
yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 question
Angelito Manansala wrote: yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Angelito Manansala www.voicefidelity.net Mobile: +639175425807 DID: (+63) 44 7906770 msn: [EMAIL PROTECTED] skype: bulcrack ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi You can give me some idea of as do it. Actaually I've the following trial network endpoint -- GK1 -- GK2 -- Asterisk GK1 configuration: Direct Mode GK2 configuration : Routed Mode Thanks in advance!! Best Regards! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 question
Angelito Manansala wrote: yes On 11/21/05, Javier Oviedo [EMAIL PROTECTED] wrote: Hi all, for h323 to h323 can asterixk *proxy **call* with *signal **only* ( no rtp go through asterisk ) thanks in advance best regards! Hello, As far as I know Asterisk cannot disentangle RTP from signaling in either SIP or H323 at least until now. I'd also be interested to know if this option is available now in case I've missed something... Best regards, Vlasis Hatzistavrou. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 one way audio using oh323
Mik, Your asterisk server is another machine of your GK ? You can start verifying if the traffic between the machines (related to RTP packets) is ok. Do you have firewall ? -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)3139-4091 / r. 108 Rio de Janeiro - Brasil www.tecnologiaip.com.br On Monday 31 October 2005 08:05, mik sib wrote: Hi all, through oh323 i can register to my gatekeeper and make and receive calls. My gatekeeper routes the incoming call as well as the outgoing. The problem is simply that i can't ear nothing from my SIP ipPhones. I can ear my voice from a normal telephone in my SIP phone but no viceversa. How can i debug this situation ? I've no errors in the log or at the asterisk startup. How to understand what's happening ? I've tryed different phones also. any idea ? thank you very much Mik Here's my oh323.conf Configuration of OpenH323 channel driver -- Version: 0.7.3 Listening on address: 10.0.0.253:1720 Gatekeeper used: [EMAIL PROTECTED] (Registered) FastStart/H245Tunnelling/H245inSetup: ON/ON/ON Supported formats in pref. order: ulaw0 Jitter buffer limits (min/max): 20-100 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: rfc2833 Max number of inbound H.323 calls: 100 Max number of outbound H.323 calls: 100 Max number of simultaneous H.323 calls: 100 Max call rate (ingress direction): 1.00/30 Default language: en Default music class: default Default context: voip-h323 doing a call with the ip phone to the outside world through the gatekeeper [2]WrapperAPI::h323_make_call: Making call. [2]WrapH323EndPoint::MakeCall: Making call to 0258115040 [4]WrapH323EndPoint::CreateConnection: Creating a H323Connection [32066] [2]WrapH323Connection::WrapH323Connection: Creation of WrapH323Connection based on user data. [2]WrapH323Connection::WrapH323Connection: Call is outgoing. [4]WrapH323Connection::WrapH323Connection: WrapH323Connection created. [3]WrapH323EndPoint::MakeCall: Call token is ip$localhost/32066 [3]WrapH323EndPoint::MakeCall: Call reference is 32066 [2]WrapH323Connection::OnSendSignalSetup: Sending SETUP message... [3]WrapH323Connection::OnSendSignalSetup: Setting display name 0432281316 Fabio Violino [3]WrapH323Connection::OnSendSignalSetup: Setting calling party number test419 [2]WrapH323Connection::OnAlerting: Ringing phone for 0258115040 ... [3]WrapH323EndPoint::OpenAudioChannel: Direction = RECODER, Buffer = 320 [2]WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8, TimeUnits 8 [2]WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 [2]WrapH323EndPoint::OpenAudioChannel: Packet size: 160 [2]WrapH323EndPoint::OpenAudioChannel: Frames per packet: 20 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k [3]WrapH323EndPoint::OpenAudioChannel: The sound channel with the application is asterisk-oh323(fd=45) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 45, mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize 160 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound channel Asterisk for recording using 1x320 byte buffers. [3]WrapH323Connection::OnEstablished: WrapH323Connection [ip$localhost/32066] established (FastStartDisabled/noH245Tunneling) [3]WrapH323EndPoint::OnConnectionEstablished: Connection [ip$localhost/32066] established. [3]WrapH323EndPoint::GetConnectionInfo: [ip$localhost/32066] RTP Media: 10.0.0.253:10004-0.0.0.0:0 [3]WrapH323EndPoint::OpenAudioChannel: Direction = PLAYER, Buffer = 320 [2]WrapH323EndPoint::OpenAudioChannel: Media format: FrameSize 8, FrameTime 8, TimeUnits 8 [2]WrapH323EndPoint::OpenAudioChannel: Codec info: FrameRate 160 [2]WrapH323EndPoint::OpenAudioChannel: Packet size: 160 [2]WrapH323EndPoint::OpenAudioChannel: Frames per packet: 20 [2]WrapH323EndPoint::OpenAudioChannel: LID Codec G.711-uLaw-64k [3]WrapH323EndPoint::OpenAudioChannel: The sound channel with the application is asterisk-oh323(fd=43) [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. [4]PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. [3]PAsteriskSoundChannel::Open: os_handle 43, mediaFormat 0, frameTime 1 ms, frameNum 20, packetSize 160 [3]WrapH323EndPoint::OpenAudioChannel: Opened sound channel Asterisk for playing using 1x320 byte buffers. [5]PAsteriskSoundChannel::Write: Written [160 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes] [4]PAsteriskSoundChannel::Read: Timeout [0 bytes]
RE:[Asterisk-Users] H323 REGISTRATION PROBLEM: Gatekeeper '[EMAIL PROTECTED] ' found but failed to register
Hi all First of all excuse me if i make such a big post, hope also to write in the right place. I need to connect my linux/asterisk (10.0.0.252) box to a Nortel PBX (192.168.1.10) with h323 I'd like to allow some phones to register via sip to asterisk and with these to the Nortel PBX wich gives me the connections to the outside world (phone) after downloading and compiling the latest asterisk source from cvs OpenH323 v1.15.6, PWlib v1.8.7 (Mimas version from Voxgratia) and oh323-0.7.3 from http://www.inaccessnetworks.com/projects/asterisk-oh323/download/asterisk-oh323-0.7.3.tar.gz starting asterisk i get snip Hi, I had the same problem in the same configuration. Asterisk finds the gatekeeper but it uses the wrong interface when it it should register. the problem is in the Mimas-patch2 release. change your pwlib to v1_9_1 and openh323 to version v1_17_2 then your registration works (again). Freddi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 and Asterisk
ooh323c installeed but do not know how to configure :( maybe googling or reading README can help woomera let me know if there's any one who has tried this. i've been testing this. It can do only alaw/ulaw and this is unusable for me. It works, but i've got some segfaults (using gnugk 2.2.2 and latest woomera) latest oh323 is the best choice for asterisk now (IMHO) :-) what i want to do it accecpt h323 calls and bill depending on the ip address and send the calls via h323 depdning on the gateway IP i add i think, that you can do this with oh323. I'm using gnugk and routng all h323 voip to oh323 Regards Kansihka ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 and Asterisk
Can you please let me know what version of oh323 you installed and the step by step process of installation. You can post your oh323.conf and extensions.conf, I should be able to help you out with your configuration. Regards, Ade. Kanishka Somaratne [EMAIL PROTECTED] wrote: hi guysI was working on asterisk and h323 for the past 2 weeksi have the following feedback please let me know if i am wrongh323 implementationI managed to install this it works, but the problem is it accecpts all calls from all ips. there is no way i can let it accecpt calls only from the IPs i give and bill depending on IPoh323 implementationmanaged to install, same as h323 implementation i can't add a list of ips and restrict access, the 729 - 723.1 codec convertion does not work well, get a robort voiceooh323cinstalleed but do not know how to configure :(woomeralet me know if there's any one who has tried this.what i want to do it accecpt h323 calls and bill depending on the ip address and send the calls via h323 depdning on the gateway IP i addRegardsKansihka ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Messenger NEW - crystal clear PC to PC calling worldwide with voicemail ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
With a liberal application of RFTW altus wrote: Good day all How do I get h323 and video working? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
RFTW or RTFM On Wed, 2005-08-10 at 09:36 -0400, Mark Phillips wrote: With a liberal application of RFTW altus wrote: Good day all How do I get h323 and video working? -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 error when trying to start Asterisk
does libpt_linux_x86_r.so.1.5.2 exist on your machine? maybe try running ldconfig or if that file is in a non-standard location, maybe add that path to ld.so.conf and then run ldconfig again On Wed, 2005-08-10 at 08:09, kurt turner wrote: Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Aug 10 09:09:18 WARNING[7824]: loader.c:440 load_modules: Loading module chan_h323.so failed! Ouch ... error while writing audio data: : Broken pipe I loading the following : Open H.323 v1.12.2, PWLib v1.5.2, expat-dev-1.95, expat-1.95, openssl-devel-0.9.6b, openssl-0.9.6b - per the read me files I did try the advsie previously give here -- http://lists.digium.com/pipermail/asterisk-users/2003-April/011019.html but that didn't work for me. I thinking I may have loaded these in the incorrect directories.. here is where they are located in (slash root) - is the following openh323 and pwlib located in root - /root/usr/src/asterisk/channels is the following - chan_h323.c - h323 - Would these being in different areas be the cause? Should I move these or remove them then reinstall them? Sorry for my noobness as I'm learning Linux for Asterisk.. I'm really a class 5 voice guy tryin to keep up!! Thanks, Kurt __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 error when trying to start Asterisk
yes.. i have the following IPD:/usr/local/lib# ls firmware libpt_linux_x86_d.so.1.5 libpt_linux_x86_r.so.1 python2.3libpt_linux_x86_d.so libpt_linux_x86_d.so.1.5.2 libpt_linux_x86_r.so.1.5libpt_linux_x86_d.so.1 libpt_linux_x86_r.so libpt_linux_x86_r.so.1.5.2 I found ldconfig under root /sbin/ldconfig when you say run ldconfig what are you saying? ldconfig -v .. right? if so I did that and I still get the h323 error listed below when firing up * anymore ideas?Derek Whitten [EMAIL PROTECTED] wrote: does libpt_linux_x86_r.so.1.5.2 exist on your machine?maybe try running ldconfig or if that file is in a non-standardlocation, maybe add that path to ld.so.conf and then run ldconfig againOn Wed, 2005-08-10 at 08:09, kurt turner wrote: Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directoryAug 10 09:09:18 WARNING[7824]: loader.c:440 load_modules: Loading module chan_h323.so failed! Ouch ... error while writing audio data: : Broken pipe I loading the following : Open H.323 v1.12.2, PWLib v1.5.2, expat-dev-1.95, expat-1.95, openssl-devel-0.9.6b, openssl-0.9.6b - per the read me files I did try the advsie previously give here -- http://lists.digium.com/pipermail/asterisk-users/2003-April/011019.html but that didn't work for me. I thinking I may have loaded these in the incorrect directories.. here is where they are located in (slash root) - is the following openh323 and pwlib located in root - /root/usr/src/asterisk/channels is the following - chan_h323.c - h323 - Would these being in different areas be the cause? Should I move these or remove them then reinstall them? Sorry for my noobness as I'm learning Linux for Asterisk.. I'm really a class 5 voice guy tryin to keep up!! Thanks, Kurt __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- -BEGIN GEEK CODE BLOCK-Version: 3.1GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w--PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK--___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323 error when trying to start Asterisk
When running ldconfig -v, did you see it find the files under the directory /usr/local/lib? If not, edit /etc/ld.so.conf with your favorite editor and add /usr/local/lib in a new line. Then rerun ldconfig -v. Check that the libpt* files were seen. kurt turner wrote: yes.. i have the following IPD:/usr/local/lib# ls firmwarelibpt_linux_x86_d.so.1.5 libpt_linux_x86_r.so.1 python2.3 libpt_linux_x86_d.solibpt_linux_x86_d.so.1.5.2 libpt_linux_x86_r.so.1.5 libpt_linux_x86_d.so.1 libpt_linux_x86_r.so libpt_linux_x86_r.so.1.5.2 I found ldconfig under root /sbin/ldconfig when you say run ldconfig what are you saying? ldconfig -v .. right? if so I did that and I still get the h323 error listed below when firing up * anymore ideas? */Derek Whitten [EMAIL PROTECTED]/* wrote: does libpt_linux_x86_r.so.1.5.2 exist on your machine? maybe try running ldconfig or if that file is in a non-standard location, maybe add that path to ld.so.conf and then run ldconfig again On Wed, 2005-08-10 at 08:09, kurt turner wrote: Asterisk has been working fine for me for several weeks using MGCP to a Adit600 for intra office calling. I have recently loaded h323 and the following errors occurs when starting asterisk. [chan_h323.so]Aug 10 09:09:18 WARNING[7824]: loader.c:258 ast_load_resource: libpt_linux_x86_r.so.1.5.2: cannot open shared object file: No such file or directory Aug 10 09:09:18 WARNING[7824]: loader.c:440 load_modules: Loading module chan_h323.so failed! Ouch ... error while writing audio data: : Broken pipe I loading the following : Open H.323 v1.12.2, PWLib v1.5.2, expat-dev-1.95, expat-1.95, openssl-devel-0.9.6b, openssl-0.9.6b - per the read me files I did try the advsie previously give here -- http://lists.digium.com/pipermail/asterisk-users/2003-April/011019.html but that didn't work for me. I thinking I may have loaded these in the incorrect directories.. here is where they are located in (slash root) - is the following openh323 and pwlib located in root - /root/usr/src/asterisk/channels is the following - chan_h323.c - h323 - Would these being in different areas be the cause? Should I move these or remove them then reinstall them? Sorry for my noobness as I'm learning Linux for Asterisk.. I'm really a class 5 voice guy tryin to keep up!! Thanks, Kurt -- JP Carballo http://www.netfone2x.com Bringing the world closer. It might look like I'm doing nothing, but at the cellular level, I'm really quite busy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
altus wrote: Good day all Im trying to get asterisk and oh323 to work I following the instruction on http://lists.digium.com/pipermail/asterisk-users/2005- January/081651.html It on fedora core 1,and I downloaded the lated dev. of asterisk Installation: tar -zxvf asterisk-oh323-0.7.1.tar.gz tar -zxvf pwlib-Janus_patch4-src-tar.gz tar -zxvf openh323-Janus_patch4-src-tar.gz cd pwlib ./configure make cd openh323 patch -p1 /root/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch (pach to openh323) cd openh323 ./configure make opt but at make opt I get this error g++: Internal error: Illegal instruction (program cc1plus) Please submit a full bug report. See URL:http://bugzilla.redhat.com/bugzilla for instructions. make[1]: *** [/root/openh323/lib/obj_linux_x86_r/h323rtp.o] Error 1 make[1]: Leaving directory `/root/openh323/src' make: *** [opt] Error 2 Can someone please help Thanks Altus Hello, oh323 how to -- http://linuxpower.blogspot.com/2005/07/h323-supports-for-asterisk.html Cheers, ~Madhawa ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323
Yes you can. Try with oh323 module: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -Mensaje original- De: altus [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register = -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
Yes, it worked here. part of oh323.conf example: . . . ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=asterisk alias=99001701 alias=99001702 . . . This defines h.323 id and the aliases for each channel. So, now I would like to know if asterisk can support h.323 gateway registration, like SIP. Can a h.323 gateway register on asterisk ? Thanks -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 10:54, Juan Salas wrote: Yes you can. Try with oh323 module: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -Mensaje original- De: altus [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register = pgpj3jix6qCO3.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323
From wiki... (http://www.voip-info.org/tiki-index.php?page=Asterisk+oh323+channels) The second option is valid only in the case where a gatekeeper is used. OH323 supports only one gatekeeper (or none, but not multiple gatekeepers). OH323 itself only acts as H.323 Gateway. As I look, asterisk didn't act like gatekeeper. JS. Yes, it worked here. part of oh323.conf example: . . . ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=asterisk alias=99001701 alias=99001702 . . . This defines h.323 id and the aliases for each channel. So, now I would like to know if asterisk can support h.323 gateway registration, like SIP. Can a h.323 gateway register on asterisk ? Thanks -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 10:54, Juan Salas wrote: Yes you can. Try with oh323 module: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -Mensaje original- De: altus [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register = -Mensaje original- De: Daniel Varella de Oliveira [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 10:42 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h323 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] h323
What is the difference? Is it like register and registrar ? If I make asterisk like a server and clients connect to it,is it a gatekway? And if I call another gateway its a gatekeeper ? On Thu, 2005-08-04 at 11:14 -0400, Juan Salas wrote: From wiki... (http://www.voip-info.org/tiki-index.php?page=Asterisk+oh323+channels) The second option is valid only in the case where a gatekeeper is used. OH323 supports only one gatekeeper (or none, but not multiple gatekeepers). OH323 itself only acts as H.323 Gateway. As I look, asterisk didn't act like gatekeeper. JS. Yes, it worked here. part of oh323.conf example: . . . ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=asterisk alias=99001701 alias=99001702 . . . This defines h.323 id and the aliases for each channel. So, now I would like to know if asterisk can support h.323 gateway registration, like SIP. Can a h.323 gateway register on asterisk ? Thanks -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 10:54, Juan Salas wrote: Yes you can. Try with oh323 module: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.html With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -Mensaje original- De: altus [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register = -Mensaje original- De: Daniel Varella de Oliveira [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 10:42 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h323 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
Thanks Juan for the information. Altus, about the gatekeeper... It acts like a DNS on the h.323 world. Defining the gatekeeper: 1. Component of an H.323 conferencing system that performs call address resolution, admission control, and subnet bandwidth management. 2. Telecommunications: H.323 entity on a LAN that provides address translation and control access to the LAN for H.323 terminals and gateways. The gatekeeper can provide other services to the H.323 terminals and gateways, such as bandwidth management and locating gateways. A gatekeeper maintains a registry of devices in the multimedia network. The devices register with the gatekeeper at startup and request admission to a call from the gatekeeper. What I am trying to make is to change traffic communication between the two protocols (SIP and H.323). And I heard that is possible using a h.323 component on the Asterisk. I'm using oh.323 component from Innaccessnetwork and until now I'm on the right way. When it will finish I will post on the list the experience. -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 12:39, altus wrote: What is the difference? Is it like register and registrar ? If I make asterisk like a server and clients connect to it,is it a gatekway? And if I call another gateway its a gatekeeper ? On Thu, 2005-08-04 at 11:14 -0400, Juan Salas wrote: From wiki... (http://www.voip-info.org/tiki-index.php?page=Asterisk+oh323+channels) The second option is valid only in the case where a gatekeeper is used. OH323 supports only one gatekeeper (or none, but not multiple gatekeepers). OH323 itself only acts as H.323 Gateway. As I look, asterisk didn't act like gatekeeper. JS. Yes, it worked here. part of oh323.conf example: . . . ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=asterisk alias=99001701 alias=99001702 . . . This defines h.323 id and the aliases for each channel. So, now I would like to know if asterisk can support h.323 gateway registration, like SIP. Can a h.323 gateway register on asterisk ? Thanks -- [ ]'s Daniel Varella de Oliveira Tecnologia IP Ltda Tel.: +55 (21)2495-0936 / r. 108 www.tecnologiaip.com.br On Thursday 04 August 2005 10:54, Juan Salas wrote: Yes you can. Try with oh323 module: http://lists.digium.com/pipermail/asterisk-users/2005-January/081881.h tml With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -Mensaje original- De: altus [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register = -Mensaje original- De: Daniel Varella de Oliveira [mailto:[EMAIL PROTECTED] Enviado el: Thursday, August 04, 2005 10:42 AM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] h323 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users pgpWdXLGBp6VW.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] h323
mercial Discussion Asunto: Re: [Asterisk-Users] h323 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users--MR Chava, Krishna SumanthGraduate Student, MEng in InternetworkingDalhousie University5562 Sackville StreetHalifax, NS, CanadaB3J 1L1email:[EMAIL PROTECTED]phone number: 1-902-440-7272__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 Configuration file
Hi, This is what I have and is working just fine. I disabled Asterisk gatekeeper and registered directly to a Cisco CallManager 3.3.4 via h323 trunk. ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Configure TCP port range to be used by H.323 ; tcpStart=1 tcpEnd=2 ; ; Configure UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; rtp.conf ; udpStart=1 udpEnd=2 ; ; Enable fast start (yes,no). ; fastStart=no ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound/simultaneous ; H.323 connections. ; outboundMax=10 inboundMax=10 simultaneousMax=10 ; ; Set the bandwidth limit for H.323 connections. ; The value is in Kbps. ; bandwidthLimit=1024 ; ; Set tracing options for the wrapper library and for the ; OpenH323 library. ; libTraceFile can be 'stdout' or a full path name to the tracefile. ; Only trace info for OpenH323 is logged in libTraceFile. ; wrapLibTraceLevel=3 libTraceLevel=3 libTraceFile=/root/h323.log ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; ;gatekeeper=192.168.2.2 gatekeeper=DISABLE AllowGKRouted=yes ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the gatekeeper registration timeout ; gatekeeperTTL=600 ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=TONE ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; ;context=voip-h323 context=from-pstn ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; alias=asterisk alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=ASTERISK alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; context=more-aliases alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; context=all-prefixes gwprefix=00 gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; ;context=from-pstn ;alias=fax ;gwprefix=14002 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726- G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G711A frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 ;codec=G729 ;frames=2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, July 27, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] H323 Configuration file Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of [EMAIL PROTECTED] installation. I have tried to use the oh323.conf content listed on WIKI but it is
