[asterisk-users] H323 Transfer Problem
Dear all; I have an incoming call from Ericsson PBX to Asterisk through H323 trunk. I need to transfer this call back to Ericsson and then Asterisk should release the channel so that if I shutdown Asterisk call should not be disconnected. As far as I know Transfer function does not work over H323 and if I use Dial command, Asterisk will remain in the path and I don't want this. Anyone knows how to do this? Any suggestion will be appreciated in advance.-- _ -- 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
[asterisk-users] H323 Transfer
Dear all; I have an incoming call from Ericsson PBX to Asterisk through H323 trunk. I need to transfer this call back to Ericsson and then Asterisk should release the channel so that if I shutdown Asterisk call should not be disconnected. As far as I know Transfer function does not work over H323 and if I use Dial command, Asterisk will remain in the path and I don't want this. Anyone knows how to do this? Any suggestion will be appreciated in advance. -- _ -- 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-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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[asterisk-users] h323-sip: one way connection
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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[asterisk-users] h323 with NAT
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 -- _ -- 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 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 --
[asterisk-users] H323 Disconnects after 15+ minutes
I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From what I can tell, the cause is condition 20 on ooh323. Any suggestions as to the cause? http://www.elastix.org/component/option,com_fireboard/Itemid,55/func,view/catid,3/id,41480/lang,en/#42715 Dec 29 10:25:01 VERBOSE [15027] logger.c: -- Remote UNIX connection Dec 29 10:25:01 VERBOSE [31438] logger.c: -- Remote UNIX connection disconnected Dec 29 10:26:01 WARNING [31413] chan_ooh323.c: Don't know how to indicate condition 20 on ooh323c_9 Dec 29 14:42:06 VERBOSE [349] logger.c: -- SIP/5034-1b1aa680 is ringing Dec 29 14:42:09 VERBOSE [349] logger.c: -- SIP/5034-1b1aa680 answered OOH323/denver-eaf3 Dec 29 14:42:09 WARNING [349] chan_ooh323.c: Don't know how to indicate condition 20 on ooh323c_18 Dec 29 15:02:55 VERBOSE [410] logger.c: -- Remote UNIX connection disconnected Dec 29 15:04:01 WARNING [349] chan_ooh323.c: Don't know how to indicate condition 20 on ooh323c_18 Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-dial:1] Macro(OOH323/denver-eaf3, hangupcall) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:1] ResetCDR(OOH323/denver-eaf3, w) in new stack Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: ResetCDR Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:2] NoCDR(OOH323/denver-eaf3, ) in new stack Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: NoCDR Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:3] GotoIf(OOH323/denver-eaf3, 1?skiprg) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,6) Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:6] GotoIf(OOH323/denver-eaf3, 1?skipblkvm) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,9) Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:9] GotoIf(OOH323/denver-eaf3, 1?theend) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: -- Goto (macro-hangupcall,s,11) Dec 29 15:04:01 DEBUG [349] app_macro.c: Executed application: GotoIf Dec 29 15:04:01 VERBOSE [349] logger.c: -- Executing [...@macro-hangupcall:11] Hangup(OOH323/denver-eaf3, ) in new stack Dec 29 15:04:01 VERBOSE [349] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'OOH323/denver-eaf3' in macro 'hangupcall' Dec 29 15:04:01 VERBOSE [349] logger.c: == Spawn h extension (macro-dial, h, 1) exited non-zero on 'OOH323/denver-eaf3' ___ -- 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 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
[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
[asterisk-users] H323 situation
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
[asterisk-users] h323 guide for asterisk
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
[asterisk-users] H323 Call Variables
Hello, Im using channel_h323 by Jeremy McNamara to connect my asterisk box to an Gatekeeper and I want to do some filter by remote ip addres but I dont know what variable in asterisk have this data. Someone knows how is the name or which are the name of this variable in channel h323? Thanks for any help! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. ggonza...@despegar.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
[asterisk-users] H323 stress test
Hello, We made small stress-test for H323. Test shows that H323 protocol is heavyweight compared with SIP. More details: http://wiki.kolmisoft.com/index.php/H323_pass-through_test Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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
[asterisk-users] H323 crashes Asterisk on high load
Hello, Asterisk 1.4.18.1 PWlib 1.10.0 Openh323 1.18.0 ../asterisk/channels/h323 compiled from source. Under high load H323 crashes and kills Asterisk, debug shows: (gdb) bt #0 0x007a2b18 in strcmp () from /lib/libc.so.6 #1 0x014478a1 in find_call_locked (call_reference=13, token=0xa1cc570 ip$81.192.72.46:7768/13) at chan_h323.c:1148 #2 0x01449f07 in cleanup_connection (call_reference=13, call_token=0xa1cc570 ip$81.192.72.46:7768/13) at chan_h323.c:2290 #3 0x0145a724 in MyH323EndPoint::OnConnectionCleared () from /usr/lib/asterisk/modules/chan_h323.so #4 0x00e604f1 in H323Connection::OnCleared () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #5 0x00e721d1 in H323EndPoint::CleanUpConnections () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #6 0x00e722fe in H323ConnectionsCleaner::Main () from /usr/local/lib/libh323_linux_x86_r.so.1.18.0 #7 0x005fd6e5 in PThread::PX_ThreadStart () from /usr/local/lib/libpt_linux_x86_r.so.1.10.0 #8 0x0088446b in start_thread () from /lib/libpthread.so.0 #9 0x00804dbe in clone () from /lib/libc.so.6 Server 2x XEON quad core and 4g DDR crashes on 110-120 simm. H323 calls. Anybody experienced same situation? Maybe there is some fix? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions ___ -- 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
[asterisk-users] H323
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
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
[asterisk-users] H323 protocol
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
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
[asterisk-users] H323 Issue
Hey, 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: deimos*CLI show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- Zap Zapata Telephony Driver w/PRIno yes no Phone Standard Linux Telephony API Driver no yes no SIP Session Initiation Protocol (SIP)yes yes yes MGCPMedia Gateway Control Protocol (MGCP)yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes Local Local Proxy Channel Driver yes yes no Feature Feature Proxy Channel Driver no yes no Console OSS Console Channel Driver no yes no Agent Call Agent Proxy Channel yes yes no Skinny Skinny Client Control Protocol (Skinny) no yes no -- 10 channel drivers registered. Has anyone experienced this before? Regards, Igor H. ___ -- 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
[asterisk-users] h323 channel compile error
I have following settings done on my Fedora8: Downloaded openh323-v1_19_0_1-src-tar.gz pwlib-v1_11_1-src.tar.gz Extracted them in /root/openh323 and /root/pwlib Exported the following variables: PWLIBDIR=/root/pwlib export PWLIBDIR OPENH323DIR=/root/openh323 export OPENH323DIR LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib export LD_LIBRARY_PATH Then I compiled pwlib and it was fine. But in compilation of openh323 as below i got the error, which failed to find solution in any forum: ./configure make Errror -- make P_SHAREDLIB=1 opt make[1]: Entering directory `/root/openh323' make -C src opt make[2]: Entering directory `/root/openh323/src' g++ -D_REENTRANT -Wall -fPIC -DPIC -DPTRACING -I/root/openh323/include -I/root/pwlib/include -Os -felide-constructors -Wreorder -c h323ep.cxx -o /root/openh323/lib/obj_linux_x86_r/h323ep.o /root/openh323/include/h4601.h: In member function ‘H460_FeatureContent::operator H460_FeatureTable*()’: /root/openh323/include/h4601.h:292: warning: type-punning to incomplete type might break strict-aliasing rules h323ep.cxx: In constructor ‘H323EndPoint::H323EndPoint()’: h323ep.cxx:1001: error: ‘PSoundChannel’ has not been declared h323ep.cxx:1001: error: ‘PSoundChannel’ has not been declared h323ep.cxx:1002: error: ‘PSoundChannel’ has not been declared h323ep.cxx:1002: error: ‘PSoundChannel’ has not been declared h323ep.cxx: In member function ‘virtual BOOL H323EndPoint::OpenAudioChannel(H323Connection, BOOL, unsigned int, H323AudioCodec)’: h323ep.cxx:2841: error: ‘PSoundChannel’ was not declared in this scope h323ep.cxx:2841: error: ‘soundChannel’ was not declared in this scope h323ep.cxx:2843: error: ‘PSoundChannel’ is not a class or namespace h323ep.cxx:2845: error: expected type-specifier before ‘PSoundChannel’ h323ep.cxx:2845: error: expected `;' before ‘PSoundChannel’ h323ep.cxx:2854: error: ‘PSoundChannel’ is not a class or namespace h323ep.cxx:2855: error: ‘PSoundChannel’ is not a class or namespace h323ep.cxx:2869: error: type ‘type error’ argument given to ‘delete’, expected pointer h323ep.cxx: In member function ‘virtual BOOL H323EndPoint::SetSoundChannelPlayDevice(const PString)’: h323ep.cxx:3047: error: ‘PSoundChannel’ has not been declared h323ep.cxx:3047: error: ‘PSoundChannel’ has not been declared h323ep.cxx: In member function ‘virtual BOOL H323EndPoint::SetSoundChannelRecordDevice(const PString)’: h323ep.cxx:3057: error: ‘PSoundChannel’ has not been declared h323ep.cxx:3057: error: ‘PSoundChannel’ has not been declared h323ep.cxx: In member function ‘virtual BOOL H323EndPoint::SetSoundChannelPlayDriver(const PString)’: h323ep.cxx:3074: error: ‘PSoundChannel’ has not been declared h323ep.cxx:3074: error: ‘PSoundChannel’ has not been declared h323ep.cxx: In member function ‘virtual BOOL H323EndPoint::SetSoundChannelRecordDriver(const PString)’: h323ep.cxx:3091: error: ‘PSoundChannel’ has not been declared h323ep.cxx:3091: error: ‘PSoundChannel’ has not been declared make[2]: *** [/root/openh323/lib/obj_linux_x86_r/h323ep.o] Error 1 make[2]: Leaving directory `/root/openh323/src' make[1]: *** [opt] Error 2 make[1]: Leaving directory `/root/openh323' make: *** [optshared] Error 2 __ I also tried make opt but the error remain same. Any idea. Please reply, Thanks Shehzad. ___ -- 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
[asterisk-users] H323 installation needed ($$$)
I am after someone to help me to config H323 on asterisk if possible since I am far too busy stuck on another project. Interested parties please msn me on sam _ _ tam AT hotmail.com please take out all space and change AT to @ If you are unsure then you can always email me with your contact via my gmail account. Sam ___ -- 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
[asterisk-users] H323 and Gatekeeper
Hi List; In the h323.conf file, the parameter gatekeeper is used to let asterisk work as h323 gatekeeper listening at port 1719 by setting gatekeeper=DISCOVER or it is used to let asterisk search for the gatekeeper to talk with it and receive calls from it? But if just to let asterisk talk with it, then what asterisk will talk with it other than receiving calls from it? Any help? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ --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 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
[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
[asterisk-users] h323 help
We've configured ooh323 on our 1.4.6 asterisk server. We've looked at various sites for tips, most recently http://www.tek-tips.com/viewthread.cfm?qid=1243330page=3. The module seems to load properly. When we do a tcpdump, we see traffic flowing between the asterisk server and the Avaya communication manager. However, we're not geting phone calls connect. Since we do not manage the Avaya CM, how can we further verify that our ooh323 config is correct? Thanks for any tips. -- Jiann-Ming Su I have to decide between two equally frightening options. If I wanted to do that, I'd vote. --Duckman The system's broke, Hank. The election baby has peed in the bath water. You got to throw 'em both out. --Dale Gribble Those who vote decide nothing. Those who count the votes decide everything. --Joseph Stalin ___ --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
[asterisk-users] H323 to H323 bridging ... failed ... also with chan_local
Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten = 111, 1, Dial(chan_h323, H323/[EMAIL PROTECTED])(in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the call to phone 2 is setup (completed) and when trying to proceed with call from phone1, asterisk stops: *CLI -- Executing Dial(H323/ip$192.168.1.100:1894/4096, H323/[EMAIL PROTECTED]) in new stack -- Called [EMAIL PROTECTED] May 7 11:29:22 WARNING[845]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/wave-1(-2033656) to H323/ip$192.168.1.100:1894/4096(-2033656) -- H323/wave-1 answered H323/ip$192.168.1.100:1894/4096 May 7 11:29:22 WARNING[845]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/ip$192.168.1.100:1894/4096(-2033656) to H323/wave-1(-2033656) May 7 11:29:22 WARNING[845]: app_dial.c:1586 dial_exec_full: Had to drop call because I couldn't make H323/ip$192.168.1.100:1894/4096 compatible with H323/wave-1 == Spawn extension (h323_default, 811, 1) exited non-zero on 'H323/ip$192.168.1.100:1894/4096' I have tried with both phones individually, and both are asterisk-compatible with H323. Bridging works if the originating call is SIP, for example. But if I try H323 with H323, it's a nono. Am I doing something wrong? do I need to set up some parameter? I thought about using chan_local, but I came across this: *CLI -- Executing Dial(H323/ip$192.168.1.100:1940/4096, local/[EMAIL PROTECTED]/n) in new stack May 7 11:31:47 WARNING[860]: channel.c:2512 ast_request: No translator path exists for channel type local (native -1) to -2033656 May 7 11:31:47 NOTICE[860]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'local' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Wait(H323/ip$192.168.1.100:1940/4096, 1) in new stack -- Executing Playback(H323/ip$192.168.1.100:1940/4096, /etc/asterisk/sounds/pbx-invalid) in new stack -- Playing '/etc/asterisk/sounds/pbx-invalid' (language 'en') Thanks in advance! Cesc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] h323 problem with asterisk 1.2.18
i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/ export OPENH323DIR LD_LIBRARY_PATH=$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
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
[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
[asterisk-users] h323 how to set it up?
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
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
[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
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
[asterisk-users] h323 - SIP conversion
I'm looking at setting up an asterisk box dedicated to SIP-H323 conversion (a 3rd party is currently converting the protocols for us). 1. Is it worthwhile to split this functionality onto a second server? Or should we let the ast pbx handle the conversion? (we have a couple hundred active channels to convert) 2. Is it better to go direct from SIP to AIX? 2. Can Asterisk handle H323 natively with problem? Thanks, MD ___ --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
[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
[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
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
[asterisk-users] h323 compile error
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
[asterisk-users] H323 NAT Problem
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
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
[asterisk-users] H323 no audio
Hi, My configuration is SipPhone-asterisk1 -asterisk2. My asterisk version is 1.2.10. I installed chan_h323 according to 'http://astrecipes.net/?n=102'. When i call from asterisk1 to asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Regards, Jason. #--h323.conf for both [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=ulaw context=default #--dial plan of asterisk1 exten = *59,1,Wait(1) exten = *59,2,Dial(H323/[EMAIL PROTECTED]) #--dial plan of asterisk2 exten = 3500,1,Playback(hello) exten = 3500,2,Hangup() #--'rtp debug' messages-- Raw PDU: 08 02 55 13 62 1c 00 7e 00 0f 05 28 10 01 00 04 ..U.b..~...( c0 01 80 05 01 03 28 00 01 ..(.. 2:15:36.845 H225 Caller:89bf340 h323.cxx(4301) H323 InternalEstablishedConnectionCheck: connectionState=EstablishedConnection fastStartState=FastStartAcknowledged Got RTP packet from 192.168.1.232:16426 (type 0, seq 1540, ts 161645797, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1541, ts 161646037, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1542, ts 161646277, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1543, ts 161646517, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1544, ts 161646757, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1545, ts 161646997, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1546, ts 161647237, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1547, ts 161647477, len 240) Got RTP packet from 192.168.1.232:16426 (type 0, seq 1548, ts 161647717, len 240) Sponsored Link Don't quit your job - take classes online www.Classesusa.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
[asterisk-users] H323 - SIP
Hi The communcation between an alcatel telephone switchbox and a sip phone (using asterisk h.323 implementation) isnt working fully bidirectional. The user at the alcatel telephone switchbox can hear the user who is speaking on the sip phone but not the other way around. Could that be a miss-configuration or a incompatibility between asterisk h.323 and pwlib/openh323? The only allowed codec is alaw and the alcatel telephone switchbox is configured as gatekeeper. Im using asterisk 1.2.12.1, pwlib 1.11.0 and openh323 1.19.0.1 Greetings Tobi -- Der GMX SmartSurfer hilft bis zu 70% Ihrer Onlinekosten zu sparen! Ideal für Modem und ISDN: http://www.gmx.net/de/go/smartsurfer ___ --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
[asterisk-users] H323 IP phones
Hi guys!Can someone give advice on nice H323 IP phones brands?? I'm looking for some H323 IP phones for a customer. Diving in theinternet found the Uniden - TVUNIDEN_UIP300, but haven't ever heard about them. Can someone give feedback experince about it??, configease, sound quality, visual appearance, end-user feedback, any infowill be appreciated.thnx!Alyed ___ --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
[asterisk-users] H323
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
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
[asterisk-users] H323 can not register to remote openh323gk?
Hi,all: in /etc/asterisk/h323.conf I setting gatekeeper=192.168.0.19 secret=3001 and on server 192.168.0.19 I running a openh323gk and add a user 3001 and password is 3001 too, but when I booting asterisk, I got messages : Error registering with gatekeeper "192.168.0.19".Aug 22 15:58:22 ERROR[2590]: chan_h323.c:2373 load_module: Gatekeeper registration failed. I don't know why? tengulre ___ --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
[asterisk-users] H323 implementation
I have a requirement to set up an Asterisk server that will handle H323. In the end this is used for video conferencing but it will be transitioning other H323 devices to SIP at some point. My question is this: Does anyone know of or have good documentation that explains how this configuration might work or should work. I understand that the implementation of H323 in Asterisk is for a gateway only. I have put GnuGK on the same box to handle the gatekeeper role and they appear to work individually but I have not tested interoperability yet (I will be later this morning). I am supposing that I just point the Asterisk gateway to the gatekeeper (which happens to be on the same box) and it should be able to handle the number mapping. The other problem I have is MCU. I did not have much luck with openMCU yet, so I am in need of that as well. I suppose this turned into a multipoint question, sorry. Has anyone done anything like this out there that was a completely capable unit that will handle (PBX functionality, PSTN connection, and MCU functionality)? Thanks Curt ___ --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
[Asterisk-Users] H323 Asterisk best practices
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 ___ --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
[Asterisk-Users] H323 to SIP Gateway
I'm trying to setup an Asterisk box as an H323 to SIP gateway. Basically, I'd like to receive traffic in H323 and forward to another Asterisk box (on the same network) using either IAX2 or SIP so that the second Asterisk box communicates with other gateways using SIP. Therefore, if I receive a request from a remote H323 gateway to dial a particular number, the H323-to-SIP gateway should forward the request to the Asterisk SIP gateway, who would simply terminate the call according to whatever rules are defined in the context. Can anyone tell me how can this be done? I setup chan_oh323 on an * box and played with the configurations but have not been able to make it all work. I can place connect the two * boxes using SIP-to-SIP as well as IAX2-to-IAX2 just fine, but have not gotten the H323 to work. Thanks, Daniel ___ --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
[Asterisk-Users] h323 phone
I installed an asterisk server with oh323 channel driver support. Then I uploaded the H323 firmware on a AT320 phone (Usually I use it as a sip phone, but I am using it just for test) Let's say that I assigned 945 as phone number, account and password to this phone, and its ip address were 192.168.1.88 Which are the right entries to add in /etc/asterisk/oh323.conf ? I tried (with no chance..) [945] type=user username=945 secret=945 host=192.168.1.88 context=from-internal incominglimit=4 thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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
[Asterisk-Users] H323 to SIP connection problem
Everyone,
I have been trying to connect a PBX with H323 IP trunks
with g711 codec to my Asterisk server running ooh323 service. I can place calls
to and from either the Asterisk, or PBX with no problem, but when I try to
pickup the call on either end, the phone hangs up immediately. Debug shows
normal to me but at the last few lines of data there is an error shown that I
have not been able to find any information to help with troubleshooting. Could
someone assist with the fix?
**Last few line from logfile**
12:48:19:257 Queued H245 messages 1. (outgoing,
ooh323c_o_2)
12:48:19:257 msgCtxt Reset? Done (outgoing,
ooh323c_o_2)
12:48:19:257 MasterSlaveDetermination done -
Slave(outgoing, ooh323c_o_2)
12:48:19:257 Not opening logical channels as Cap
exchange remaining
12:48:19:257 Finished handling H245 message.
(outgoing, ooh323c_o_2)
12:48:19:257 Receiving H.2250 message (outgoing,
ooh323c_o_2)
12:48:19:257 Received Q.931 message: (outgoing,
ooh323c_o_2)
12:48:19:257 Received H.2250 Message = {
12:48:19:258 protocolDiscriminator
= 8
12:48:19:258 callReference = 43
12:48:19:258 from = destination
12:48:19:258 messageType = 5a
12:48:19:258 Cause IE = {
12:48:19:258
Unsupported Cause Type
12:48:19:258 }
12:48:19:258 h323_uu_pdu = {
12:48:19:258
h323_message_body = {
12:48:19:258
releaseComplete = {
12:48:19:259
protocolIdentifier = {
12:48:19:259
{
12:48:19:260 0 0 8 2250 0 2 }
12:48:19:261
}
12:48:19:261
callIdentifier = {
12:48:19:262
guid = {
12:48:19:262
'6f6f68333233632d818e86c6'H
12:48:19:263
}
12:48:19:264
}
12:48:19:264
}
12:48:19:265 }
12:48:19:265 }
12:48:19:265 UUIE decode successful
12:48:19:265 Decoded Q931 message (outgoing,
ooh323c_o_2)
12:48:19:265 }
12:48:19:265 H.225 Release Complete message received
(outgoing, ooh323c_o_2)
12:48:19:265 Cause of Release Complete is 0.
(outgoing, ooh323c_o_2)
12:48:19:265 Closing H.245 connection (outgoing,
ooh323c_o_2)
12:48:19:266 Closed H245 connection. (outgoing,
ooh323c_o_2)
12:48:19:266 In ooEndCall call state is -
OO_CALL_CLEARED (outgoing, ooh323c_o_2)
12:48:19:266 Cleaning Call (outgoing, ooh323c_o_2)-
reason:OO_REASON_UNKNOWN
12:48:19:266 Removed call (outgoing, ooh323c_o_2)
from list
DJ
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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
[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 ___ --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
[Asterisk-Users] h323 with asterisk problem
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 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
[Asterisk-Users] h323 to sip ringing indication
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 ___ --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
[Asterisk-Users] H323 calls will not stay connected
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? _ Richard wrote: Try restricting both ends to one codec;disallow=allallow=codec of choiceat the asterisk end and whatever you need to do at the legacy end.Regards,Richard Daren J. Howell [EMAIL PROTECTED] www.dtcommunication.com PH 678.388.9163 FX 678.921.2133 ___ --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
[Asterisk-Users] H323 calls will not stay connected
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.
No gatekeeper is installed. I have attached a copy of my
h323 logfile for debugging.
What do you suggest what change needs to take place to keep
calls connected?
11:33:19:864 Queued H245 messages 1. (incoming,
ooh323c_7)
11:33:19:864 msgCtxt Reset? Done (incoming,
ooh323c_7)
11:33:19:864 MasterSlaveDetermination done -
Slave(incoming, ooh323c_7)
11:33:19:864 Not opening logical channels as Cap
exchange remaining
11:33:19:864 Finished handling H245 message.
(incoming, ooh323c_7)
11:33:19:864 Receiving H.2250 message (incoming,
ooh323c_7)
11:33:19:864 Received Q.931 message: (incoming,
ooh323c_7)
11:33:19:864 Received H.2250 Message = {
11:33:19:864 protocolDiscriminator
= 8
11:33:19:864 callReference = 2
11:33:19:865 from = originator
11:33:19:865 messageType = 5a
11:33:19:865 Cause IE = {
11:33:19:865
Unsupported Cause Type
11:33:19:865 }
11:33:19:865 h323_uu_pdu = {
11:33:19:865
h323_message_body = {
11:33:19:865
releaseComplete = {
11:33:19:866
protocolIdentifier = {
11:33:19:866
{
11:33:19:867 0 0 8 2250 0 2 }
11:33:19:867
}
11:33:19:868
callIdentifier = {
11:33:19:868
guid = {
11:33:19:869
'0002010507d6080b21380ef4016a'H
11:33:19:870
}
11:33:19:870
}
11:33:19:871
}
11:33:19:871 }
11:33:19:872 }
11:33:19:872 UUIE decode successful
11:33:19:872 Decoded Q931 message (incoming,
ooh323c_7)
11:33:19:872 }
11:33:19:872 H.225 Release Complete message received
(incoming, ooh323c_7)
11:33:19:872 Cause of Release Complete is 0.
(incoming, ooh323c_7)
11:33:19:873 Closing H.245 connection (incoming,
ooh323c_7)
11:33:19:873 Closed H245 connection. (incoming,
ooh323c_7)
11:33:19:873 In ooEndCall call state is -
OO_CALL_CLEARED (incoming, ooh323c_7)
11:33:19:873 Cleaning Call (incoming, ooh323c_7)-
reason:OO_REASON_UNKNOWN
11:33:19:873 Closing H.245 Listener (incoming,
ooh323c_7)
11:33:19:873 Removed call (incoming, ooh323c_7) from
list
[EMAIL PROTECTED] ~]#
DJ.
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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
[Asterisk-Users] H323 to SIP
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
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 dont 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
