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[Freeswitch-users] mod_opal first unsuccessful test


 
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regs at kinetix.gr
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PostPosted: Mon Jan 12, 2009 11:03 am    Post subject: [Freeswitch-users] mod_opal first unsuccessful test Reply with quote

Hi,

I successfully compiled mod_opal using the latest svn for both opal
and ptlib as Brian suggested.

When I try to establish a call using h323 from my openphone client
I get no audio although I can see RTP packets in both directions when I am
doing a capture. Some details :

I am using the 11094 revision of the FS trunk.
I am using the PCMU codec.
I tried dialing the default IVR (5000) and other testing extensions
(freeswitch conference, echo test etc.)
I tried using fast start on and off , h245 tunneling on and off, h245 in
SETUP on and off.

In my captures I have also noticed a strange behavior : FS sends to
my client 2 "alerting" packets
for no apparent reason. Could this be a cause of the problem?

Had anyone any success with mod_opal lately? If yes, could you
please reply quoting your config
options (both on FS and on your client)?


--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
-------------------------------------------


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Back to top
anthony.minessale at g...
Guest





PostPosted: Mon Jan 12, 2009 2:20 pm    Post subject: [Freeswitch-users] mod_opal first unsuccessful test Reply with quote

I have only tested it with x-meeting so far.

On Mon, Jan 12, 2009 at 9:51 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> wrote:
Quote:
Hi,

I successfully compiled mod_opal using the latest svn for both opal
and ptlib as Brian suggested.

When I try to establish a call using h323 from my openphone client
I get no audio although I can see RTP packets in both directions when I am
doing a capture. Some details :

I am using the 11094 revision of the FS trunk.
I am using the PCMU codec.
I tried dialing the default IVR (5000) and other testing extensions
(freeswitch conference, echo test etc.)
I tried using fast start on and off , h245 tunneling on and off, h245 in
SETUP on and off.

In my captures I have also noticed a strange behavior : FS sends to
my client 2 "alerting" packets
for no apparent reason. Could this be a cause of the problem?

Had anyone any success with mod_opal lately? If yes, could you
please reply quoting your config
options (both on FS and on your client)?


--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
Back to top
regs at kinetix.gr
Guest





PostPosted: Tue Jan 13, 2009 6:51 am    Post subject: [Freeswitch-users] mod_opal first unsuccessful test Reply with quote

Hi,

Yes, openPhone is working with my soundcard. I am using it
every day for testing purposes. I use the 1.8.1 version. Is there a newer
version that uses OPAL? I didn't know that. Where can I get it from?

Robert Jongbloed wrote:
Quote:
<![endif]--> <![endif]-->
Hi guys,

I was using the OpenPhone that you build with OPAL for my testing. So that is identical (I think) to you.

I have not (yet) do any third party client testing.

Two ALERTING messages are fine, perfectly legal and OPAL can handle it.

You say you can see the RTP packets flowing so that implies that the mod_opal is actually working, so let’s look somewhere else. Have you confirmed that OpenPhone is using the sound card correctly? Made a call between two machines JUST using OpenPhone for example?


Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.

From: Anthony Minessale [mailto:anthony.minessale@gmail.com (anthony.minessale@gmail.com)]
Sent: Tuesday, 13 January 2009 6:20 AM
To: Robert Jongbloed
Subject: Fwd: [Freeswitch-users] mod_opal first unsuccessful test



Heh,
what client are you using in your tests that are working?


---------- Forwarded message ----------
From: Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)>
Date: Mon, Jan 12, 2009 at 9:51 AM
Subject: [Freeswitch-users] mod_opal first unsuccessful test
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)


Hi,

I successfully compiled mod_opal using the latest svn for both opal
and ptlib as Brian suggested.

When I try to establish a call using h323 from my openphone client
I get no audio although I can see RTP packets in both directions when I am
doing a capture. Some details :

I am using the 11094 revision of the FS trunk.
I am using the PCMU codec.
I tried dialing the default IVR (5000) and other testing extensions
(freeswitch conference, echo test etc.)
I tried using fast start on and off , h245 tunneling on and off, h245 in
SETUP on and off.

In my captures I have also noticed a strange behavior : FS sends to
my client 2 "alerting" packets
for no apparent reason. Could this be a cause of the problem?

Had anyone any success with mod_opal lately? If yes, could you
please reply quoting your config
options (both on FS and on your client)?


--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------
Back to top
regs at kinetix.gr
Guest





PostPosted: Tue Jan 13, 2009 7:07 am    Post subject: [Freeswitch-users] mod_opal first unsuccessful test Reply with quote

I also tried using Ekiga - which is OPAL based - and got the same
behavior. No audio - although I can see RTP packets.

Apostolos Pantsiopoulos wrote:
Quote:
Hi,

Yes, openPhone is working with my soundcard. I am using it
every day for testing purposes. I use the 1.8.1 version. Is there a newer
version that uses OPAL? I didn't know that. Where can I get it from?

Robert Jongbloed wrote:
Quote:
<![endif]--> <![endif]-->
Hi guys,

I was using the OpenPhone that you build with OPAL for my testing. So that is identical (I think) to you.

I have not (yet) do any third party client testing.

Two ALERTING messages are fine, perfectly legal and OPAL can handle it.

You say you can see the RTP packets flowing so that implies that the mod_opal is actually working, so let’s look somewhere else. Have you confirmed that OpenPhone is using the sound card correctly? Made a call between two machines JUST using OpenPhone for example?


Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.

From: Anthony Minessale [mailto:anthony.minessale@gmail.com (anthony.minessale@gmail.com)]
Sent: Tuesday, 13 January 2009 6:20 AM
To: Robert Jongbloed
Subject: Fwd: [Freeswitch-users] mod_opal first unsuccessful test



Heh,
what client are you using in your tests that are working?


---------- Forwarded message ----------
From: Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)>
Date: Mon, Jan 12, 2009 at 9:51 AM
Subject: [Freeswitch-users] mod_opal first unsuccessful test
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)


Hi,

I successfully compiled mod_opal using the latest svn for both opal
and ptlib as Brian suggested.

When I try to establish a call using h323 from my openphone client
I get no audio although I can see RTP packets in both directions when I am
doing a capture. Some details :

I am using the 11094 revision of the FS trunk.
I am using the PCMU codec.
I tried dialing the default IVR (5000) and other testing extensions
(freeswitch conference, echo test etc.)
I tried using fast start on and off , h245 tunneling on and off, h245 in
SETUP on and off.

In my captures I have also noticed a strange behavior : FS sends to
my client 2 "alerting" packets
for no apparent reason. Could this be a cause of the problem?

Had anyone any success with mod_opal lately? If yes, could you
please reply quoting your config
options (both on FS and on your client)?


--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr
-------------------------------------------

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------
Back to top
regs at kinetix.gr
Guest





PostPosted: Fri Jan 16, 2009 5:37 am    Post subject: [Freeswitch-users] mod_opal first unsuccessful test Reply with quote

Any news regarding this issue?

Apostolos Pantsiopoulos wrote:
Quote:
I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and
FS is on xxx.xxx.xxx.212

Robert Jongbloed wrote:
Quote:
v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);} <![endif]--> <![endif]--> <![endif]-->
Can you send me a WireShark capture?

Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.


From: Apostolos Pantsiopoulos [mailto:regs@kinetix.gr (regs@kinetix.gr)]
Sent: Tuesday, 13 January 2009 11:05 PM
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Cc: Robert Jongbloed
Subject: Re: [Freeswitch-users] mod_opal first unsuccessful test



I also tried using Ekiga - which is OPAL based - and got the same
behavior. No audio - although I can see RTP packets.

Apostolos Pantsiopoulos wrote:
Hi,

Yes, openPhone is working with my soundcard. I am using it
every day for testing purposes. I use the 1.8.1 version. Is there a newer
version that uses OPAL? I didn't know that. Where can I get it from?

Robert Jongbloed wrote:
Hi guys,

I was using the OpenPhone that you build with OPAL for my testing. So that is identical (I think) to you.

I have not (yet) do any third party client testing.

Two ALERTING messages are fine, perfectly legal and OPAL can handle it.

You say you can see the RTP packets flowing so that implies that the mod_opal is actually working, so let’s look somewhere else. Have you confirmed that OpenPhone is using the sound card correctly? Made a call between two machines JUST using OpenPhone for example?


Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.

From: Anthony Minessale [mailto:anthony.minessale@gmail.com (anthony.minessale@gmail.com)]
Sent: Tuesday, 13 January 2009 6:20 AM
To: Robert Jongbloed
Subject: Fwd: [Freeswitch-users] mod_opal first unsuccessful test



Heh,
what client are you using in your tests that are working?



---------- Forwarded message ----------
From: Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)>
Date: Mon, Jan 12, 2009 at 9:51 AM
Subject: [Freeswitch-users] mod_opal first unsuccessful test
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)


Hi,

I successfully compiled mod_opal using the latest svn for both opal
and ptlib as Brian suggested.

When I try to establish a call using h323 from my openphone client
I get no audio although I can see RTP packets in both directions when I am
doing a capture. Some details :

I am using the 11094 revision of the FS trunk.
I am using the PCMU codec.
I tried dialing the default IVR (5000) and other testing extensions
(freeswitch conference, echo test etc.)
I tried using fast start on and off , h245 tunneling on and off, h245 in
SETUP on and off.

In my captures I have also noticed a strange behavior : FS sends to
my client 2 "alerting" packets
for no apparent reason. Could this be a cause of the problem?

Had anyone any success with mod_opal lately? If yes, could you
please reply quoting your config
options (both on FS and on your client)?


--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400




Quote:
-- -------------------------------------------Apostolos PantsiopoulosKinetix Tele.com R & Demail: regs@kinetix.gr (regs@kinetix.gr)-------------------------------------------  

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Apostolos Pantsiopoulos
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3
Back to top
regs at kinetix.gr
Guest





PostPosted: Mon Jan 19, 2009 8:59 am    Post subject: [Freeswitch-users] mod_opal first unsuccessful test Reply with quote

Sorry. I am attaching it know

Robert Jongbloed wrote:
Quote:
v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);} w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);} <![endif]--> <![endif]--> <![endif]-->
Um, there is nothing attached ....

Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.


From: Apostolos Pantsiopoulos [mailto:regs@kinetix.gr (regs@kinetix.gr)]
Sent: Friday, 16 January 2009 9:33 PM
To: Robert Jongbloed; freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Subject: Re: [Freeswitch-users] mod_opal first unsuccessful test



Any news regarding this issue?

Apostolos Pantsiopoulos wrote:
I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and
FS is on xxx.xxx.xxx.212

Robert Jongbloed wrote:
Can you send me a WireShark capture?

Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.


From: Apostolos Pantsiopoulos [mailto:regs@kinetix.gr (regs@kinetix.gr)]
Sent: Tuesday, 13 January 2009 11:05 PM
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Cc: Robert Jongbloed
Subject: Re: [Freeswitch-users] mod_opal first unsuccessful test



I also tried using Ekiga - which is OPAL based - and got the same
behavior. No audio - although I can see RTP packets.

Apostolos Pantsiopoulos wrote:
Hi,

Yes, openPhone is working with my soundcard. I am using it
every day for testing purposes. I use the 1.8.1 version. Is there a newer
version that uses OPAL? I didn't know that. Where can I get it from?

Robert Jongbloed wrote:
Hi guys,

I was using the OpenPhone that you build with OPAL for my testing. So that is identical (I think) to you.

I have not (yet) do any third party client testing.

Two ALERTING messages are fine, perfectly legal and OPAL can handle it.

You say you can see the RTP packets flowing so that implies that the mod_opal is actually working, so let’s look somewhere else. Have you confirmed that OpenPhone is using the sound card correctly? Made a call between two machines JUST using OpenPhone for example?


Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.

From: Anthony Minessale [mailto:anthony.minessale@gmail.com (anthony.minessale@gmail.com)]
Sent: Tuesday, 13 January 2009 6:20 AM
To: Robert Jongbloed
Subject: Fwd: [Freeswitch-users] mod_opal first unsuccessful test



Heh,
what client are you using in your tests that are working?




---------- Forwarded message ----------
From: Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)>
Date: Mon, Jan 12, 2009 at 9:51 AM
Subject: [Freeswitch-users] mod_opal first unsuccessful test
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)


Hi,

I successfully compiled mod_opal using the latest svn for both opal
and ptlib as Brian suggested.

When I try to establish a call using h323 from my openphone client
I get no audio although I can see RTP packets in both directions when I am
doing a capture. Some details :

I am using the 11094 revision of the FS trunk.
I am using the PCMU codec.
I tried dialing the default IVR (5000) and other testing extensions
(freeswitch conference, echo test etc.)
I tried using fast start on and off , h245 tunneling on and off, h245 in
SETUP on and off.

In my captures I have also noticed a strange behavior : FS sends to
my client 2 "alerting" packets
for no apparent reason. Could this be a cause of the problem?

Had anyone any success with mod_opal lately? If yes, could you
please reply quoting your config
options (both on FS and on your client)?


--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400





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-- -------------------------------------------Apostolos PantsiopoulosKinetix Tele.com R & Demail: regs@kinetix.gr (regs@kinetix.gr)-------------------------------------------  


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anthony.minessale at g...
Guest





PostPosted: Mon Jan 19, 2009 9:07 am    Post subject: [Freeswitch-users] mod_opal first unsuccessful test Reply with quote

robert is not on this email list so I never see his replies nor does he probably see mine.
Can you please move this thread to be between our 3 email addresses directly?


On Mon, Jan 19, 2009 at 7:57 AM, Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)> wrote:
Quote:
Sorry. I am attaching it know

Robert Jongbloed wrote:
Quote:

Um, there is nothing attached ....

Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.



From: Apostolos Pantsiopoulos [mailto:regs@kinetix.gr (regs@kinetix.gr)]
Sent: Friday, 16 January 2009 9:33 PM
To: Robert Jongbloed; freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)

Subject: Re: [Freeswitch-users] mod_opal first unsuccessful test






Any news regarding this issue?

Apostolos Pantsiopoulos wrote:
I am attaching the wireshark capture. Openphone is on xxx.xxx.xxx.202 and
FS is on xxx.xxx.xxx.212

Robert Jongbloed wrote:
Can you send me a WireShark capture?

Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.


From: Apostolos Pantsiopoulos [mailto:regs@kinetix.gr (regs@kinetix.gr)]
Sent: Tuesday, 13 January 2009 11:05 PM
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Cc: Robert Jongbloed
Subject: Re: [Freeswitch-users] mod_opal first unsuccessful test



I also tried using Ekiga - which is OPAL based - and got the same
behavior. No audio - although I can see RTP packets.

Apostolos Pantsiopoulos wrote:
Hi,

Yes, openPhone is working with my soundcard. I am using it
every day for testing purposes. I use the 1.8.1 version. Is there a newer
version that uses OPAL? I didn't know that. Where can I get it from?

Robert Jongbloed wrote:
Hi guys,

I was using the OpenPhone that you build with OPAL for my testing. So that is identical (I think) to you.

I have not (yet) do any third party client testing.

Two ALERTING messages are fine, perfectly legal and OPAL can handle it.

You say you can see the RTP packets flowing so that implies that the mod_opal is actually working, so let's look somewhere else. Have you confirmed that OpenPhone is using the sound card correctly? Made a call between two machines JUST using OpenPhone for example?


Robert Jongbloed
OPAL/OpenH323/PTLib Architect and Co-founder.

From: Anthony Minessale [mailto:anthony.minessale@gmail.com (anthony.minessale@gmail.com)]
Sent: Tuesday, 13 January 2009 6:20 AM
To: Robert Jongbloed
Subject: Fwd: [Freeswitch-users] mod_opal first unsuccessful test



Heh,
what client are you using in your tests that are working?




---------- Forwarded message ----------
From: Apostolos Pantsiopoulos <regs@kinetix.gr (regs@kinetix.gr)>
Date: Mon, Jan 12, 2009 at 9:51 AM
Subject: [Freeswitch-users] mod_opal first unsuccessful test
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)


Hi,

I successfully compiled mod_opal using the latest svn for both opal
and ptlib as Brian suggested.

When I try to establish a call using h323 from my openphone client
I get no audio although I can see RTP packets in both directions when I am
doing a capture. Some details :

I am using the 11094 revision of the FS trunk.
I am using the PCMU codec.
I tried dialing the default IVR (5000) and other testing extensions
(freeswitch conference, echo test etc.)
I tried using fast start on and off , h245 tunneling on and off, h245 in
SETUP on and off.

In my captures I have also noticed a strange behavior : FS sends to
my client 2 "alerting" packets
for no apparent reason. Could this be a cause of the problem?

Had anyone any success with mod_opal lately? If yes, could you
please reply quoting your config
options (both on FS and on your client)?


--
-------------------------------------------
Apostolos Pantsiopoulos
Kinetix Tele.com R & D
email: regs@kinetix.gr (regs@kinetix.gr)
-------------------------------------------


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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

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pstn:213-799-1400





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http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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