[sip-comm-dev] One-Way Audio Problem


#1

Hi All,

We are experiencing �one-way� audio problem between two sip-communicator clients.
Both clients register successfully with the asterisk server. The clients and server are all on the same LAN. There is no firewall, or router in the test configuration:

When Client 1111 initiates a call to 1114:
     - 1111 displays �alerting �� and generates a ringing tone
     - 1114 displays �incoming..� and generates a ringing tone
When Client 1114 accepts the call:
     - 1111 continues to ring for 2 secs, and displays �connected�
     - 1114 status display changes from �incoming� to �connecting�
When 1111 speaks:
     - 1114 clearly hears 1111
     - But 1114 status continues to display �connecting�
When 1114 speaks:
    - 1114 voice is echoed in 1114�s earpiece
    - 1111 does not hear 1114
    - 1114 status continues to display �connecting�

Test environment:
  - sip-communicator-1.0-alpha1-src.zip [also tried with nightly build of 6-June ]
  - apache-ant-1.7.0
  - jdk1.5.0_12 [ also tried the latest version 1.6]
  - jre1.5.0-12 [ also tried 1.6]
  - The sip-communicator clients run on Windows OS
  - The asterisk server runs on Debian Linux

The screen Capture for ckient 1114 is attached

We would appreciate your timely assistance

Thanks,
Gus

Client-1114 Screen Capture.doc (60.5 KB)

···

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#2

Hi,

I have been tracking this last week and the problem is that asterisk doesn't send the last ACK message to sip-communicator thats why it stays in state
"connecting".

Gus Samba wrote:

···

Hi All,

We are experiencing �one-way� audio problem between two sip-communicator clients.
Both clients register successfully with the asterisk server. The clients and server are all on the same LAN. There is no firewall, or router in the test configuration:

When Client 1111 initiates a call to 1114:
- 1111 displays �alerting �� and generates a ringing tone
- 1114 displays �incoming..� and generates a ringing tone
When Client 1114 accepts the call:
- 1111 continues to ring for 2 secs, and displays �connected�
- 1114 status display changes from �incoming� to �connecting�
When 1111 speaks:
- 1114 clearly hears 1111
- But 1114 status continues to display �connecting�
When 1114 speaks:
- 1114 voice is echoed in 1114�s earpiece
- 1111 does not hear 1114
- 1114 status continues to display �connecting�

Test environment:
- sip-communicator-1.0-alpha1-src.zip [also tried with nightly build of 6-June ]
- apache-ant-1.7.0
- jdk1.5.0_12 [ also tried the latest version 1.6]
- jre1.5.0-12 [ also tried 1.6]
- The sip-communicator clients run on Windows OS
- The asterisk server runs on Debian Linux

The screen Capture for ckient 1114 is attached

We would appreciate your timely assistance

Thanks,
Gus

_________________________________________________________________
Get a preview of Live Earth, the hottest event this summer - only on MSN http://liveearth.msn.com?source=msntaglineliveearthhm
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#3

Can you please tell me wich Asterisk version are you using?? I'm working
with asterisk-1.2.18.

···

On 6/9/07, Gus Samba <gus_samba@hotmail.com> wrote:

Hi All,

We are experiencing "one-way" audio problem between two sip-communicator
clients.
Both clients register successfully with the asterisk server. The clients
and
server are all on the same LAN. There is no firewall, or router in the
test
configuration:

When Client 1111 initiates a call to 1114:
     - 1111 displays "alerting …" and generates a ringing tone
     - 1114 displays "incoming.." and generates a ringing tone
When Client 1114 accepts the call:
     - 1111 continues to ring for 2 secs, and displays "connected"
     - 1114 status display changes from "incoming" to "connecting"
When 1111 speaks:
     - 1114 clearly hears 1111
     - But 1114 status continues to display "connecting"
When 1114 speaks:
    - 1114 voice is echoed in 1114's earpiece
    - 1111 does not hear 1114
    - 1114 status continues to display "connecting"

Test environment:
  - sip-communicator-1.0-alpha1-src.zip [also tried with nightly build of
6-June ]
  - apache-ant-1.7.0
  - jdk1.5.0_12 [ also tried the latest version 1.6]
  - jre1.5.0-12 [ also tried 1.6]
  - The sip-communicator clients run on Windows OS
  - The asterisk server runs on Debian Linux

The screen Capture for ckient 1114 is attached

We would appreciate your timely assistance

Thanks,
Gus

_________________________________________________________________
Get a preview of Live Earth, the hottest event this summer - only on MSN
http://liveearth.msn.com?source=msntaglineliveearthhm

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#4

Hi,

hum. I've tested it with the latest 1.4 version. I will test it now with 1.2.12.1 as this I have now - and will report it. But tell me when the problem appears with or without NAT ?

damencho

asmouta wrote:

···

Can you please tell me wich Asterisk version are you using?? I'm working with asterisk-1.2.18.

On 6/9/07, *Gus Samba* < gus_samba@hotmail.com > <mailto:gus_samba@hotmail.com>> wrote:

    Hi All,

    We are experiencing "one-way" audio problem between two
    sip-communicator
    clients.
    Both clients register successfully with the asterisk server. The
    clients and
    server are all on the same LAN. There is no firewall, or router
    in the test
    configuration:

    When Client 1111 initiates a call to 1114:
         - 1111 displays "alerting �" and generates a ringing tone
         - 1114 displays "incoming.." and generates a ringing tone
    When Client 1114 accepts the call:
         - 1111 continues to ring for 2 secs, and displays "connected"
         - 1114 status display changes from "incoming" to "connecting"
    When 1111 speaks:
         - 1114 clearly hears 1111
         - But 1114 status continues to display "connecting"
    When 1114 speaks:
        - 1114 voice is echoed in 1114's earpiece
        - 1111 does not hear 1114
        - 1114 status continues to display "connecting"

    Test environment:
      - sip-communicator-1.0-alpha1-src.zip [also tried with nightly
    build of
    6-June ]
      - apache-ant-1.7.0
      - jdk1.5.0_12 [ also tried the latest version 1.6]
      - jre1.5.0-12 [ also tried 1.6]
      - The sip-communicator clients run on Windows OS
      - The asterisk server runs on Debian Linux

    The screen Capture for ckient 1114 is attached

    We would appreciate your timely assistance

    Thanks,
    Gus

    _________________________________________________________________
    Get a preview of Live Earth, the hottest event this summer - only
    on MSN
    http://liveearth.msn.com?source=msntaglineliveearthhm

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#5

hi again,

to me with asterisk 1.2... works ok - with no NAT between asterisk and sip-communicator.

damencho

asmouta wrote:

···

Can you please tell me wich Asterisk version are you using?? I'm working with asterisk-1.2.18.

On 6/9/07, *Gus Samba* < gus_samba@hotmail.com > <mailto:gus_samba@hotmail.com>> wrote:

    Hi All,

    We are experiencing "one-way" audio problem between two
    sip-communicator
    clients.
    Both clients register successfully with the asterisk server. The
    clients and
    server are all on the same LAN. There is no firewall, or router
    in the test
    configuration:

    When Client 1111 initiates a call to 1114:
         - 1111 displays "alerting �" and generates a ringing tone
         - 1114 displays "incoming.." and generates a ringing tone
    When Client 1114 accepts the call:
         - 1111 continues to ring for 2 secs, and displays "connected"
         - 1114 status display changes from "incoming" to "connecting"
    When 1111 speaks:
         - 1114 clearly hears 1111
         - But 1114 status continues to display "connecting"
    When 1114 speaks:
        - 1114 voice is echoed in 1114's earpiece
        - 1111 does not hear 1114
        - 1114 status continues to display "connecting"

    Test environment:
      - sip-communicator-1.0-alpha1-src.zip [also tried with nightly
    build of
    6-June ]
      - apache-ant-1.7.0
      - jdk1.5.0_12 [ also tried the latest version 1.6]
      - jre1.5.0-12 [ also tried 1.6]
      - The sip-communicator clients run on Windows OS
      - The asterisk server runs on Debian Linux

    The screen Capture for ckient 1114 is attached

    We would appreciate your timely assistance

    Thanks,
    Gus

    _________________________________________________________________
    Get a preview of Live Earth, the hottest event this summer - only
    on MSN
    http://liveearth.msn.com?source=msntaglineliveearthhm

    ---------------------------------------------------------------------
    To unsubscribe, e-mail:
    dev-unsubscribe@sip-communicator.dev.java.net
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#6

Hi and sorry for delay, I was doing some captures.. In fact, I have no NAT
or firewall.
I'm doing tests over a LAN, two pcs (Mandriva 2006 and Suse) connected with
a modem-router the F@st1500. In the 192.168.27.105 I've installed the
Asterisk and the SIP client 1561, and in the 192.168.27.104, the 1560 SIP
client.
I'm testing with the source of the first release "alpha-1" but I've also
made tests with the windows version.

I joined with this mail two captures, in the first the 1561 calls the 1560
and the second you can find the other way call.

The behaviour is the same, the callee stay in the status "connecting" while
the caller gets the status "connected" even if the callee terminate the
call.

Thanks a lot for help.

EtherealCapture.txt (4.65 KB)

EtherealCapture2.txt (371 KB)

···

On 6/14/07, Damian Minkov <damencho@damencho.com> wrote:

Hi,

hum. I've tested it with the latest 1.4 version. I will test it now with
1.2.12.1 as this I have now - and will report it. But tell me when the
problem appears with or without NAT ?

damencho

asmouta wrote:
> Can you please tell me wich Asterisk version are you using?? I'm
> working with asterisk-1.2.18.
>
> On 6/9/07, *Gus Samba* < gus_samba@hotmail.com > > <mailto:gus_samba@hotmail.com>> wrote:
>
> Hi All,
>
> We are experiencing "one-way" audio problem between two
> sip-communicator
> clients.
> Both clients register successfully with the asterisk server. The
> clients and
> server are all on the same LAN. There is no firewall, or router
> in the test
> configuration:
>
> When Client 1111 initiates a call to 1114:
> - 1111 displays "alerting …" and generates a ringing tone
> - 1114 displays "incoming.." and generates a ringing tone
> When Client 1114 accepts the call:
> - 1111 continues to ring for 2 secs, and displays "connected"
> - 1114 status display changes from "incoming" to "connecting"
> When 1111 speaks:
> - 1114 clearly hears 1111
> - But 1114 status continues to display "connecting"
> When 1114 speaks:
> - 1114 voice is echoed in 1114's earpiece
> - 1111 does not hear 1114
> - 1114 status continues to display "connecting"
>
> Test environment:
> - sip-communicator-1.0-alpha1-src.zip [also tried with nightly
> build of
> 6-June ]
> - apache-ant-1.7.0
> - jdk1.5.0_12 [ also tried the latest version 1.6]
> - jre1.5.0-12 [ also tried 1.6]
> - The sip-communicator clients run on Windows OS
> - The asterisk server runs on Debian Linux
>
> The screen Capture for ckient 1114 is attached
>
> We would appreciate your timely assistance
>
> Thanks,
> Gus
>
> _________________________________________________________________
> Get a preview of Live Earth, the hottest event this summer - only
> on MSN
> http://liveearth.msn.com?source=msntaglineliveearthhm
>
---------------------------------------------------------------------
> To unsubscribe, e-mail:
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> <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
> For additional commands, e-mail:
> dev-help@sip-communicator.dev.java.net
> <mailto:dev-help@sip-communicator.dev.java.net>
>

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#7

Can you please send me your SIP configuration for Asterisk? I dont
understand what's happening :frowning:

···

On 6/14/07, Damian Minkov <damencho@damencho.com> wrote:

hi again,

to me with asterisk 1.2... works ok - with no NAT between asterisk and
sip-communicator.

damencho

asmouta wrote:
> Can you please tell me wich Asterisk version are you using?? I'm
> working with asterisk-1.2.18.
>
> On 6/9/07, *Gus Samba* < gus_samba@hotmail.com > > <mailto:gus_samba@hotmail.com>> wrote:
>
> Hi All,
>
> We are experiencing "one-way" audio problem between two
> sip-communicator
> clients.
> Both clients register successfully with the asterisk server. The
> clients and
> server are all on the same LAN. There is no firewall, or router
> in the test
> configuration:
>
> When Client 1111 initiates a call to 1114:
> - 1111 displays "alerting …" and generates a ringing tone
> - 1114 displays "incoming.." and generates a ringing tone
> When Client 1114 accepts the call:
> - 1111 continues to ring for 2 secs, and displays "connected"
> - 1114 status display changes from "incoming" to "connecting"
> When 1111 speaks:
> - 1114 clearly hears 1111
> - But 1114 status continues to display "connecting"
> When 1114 speaks:
> - 1114 voice is echoed in 1114's earpiece
> - 1111 does not hear 1114
> - 1114 status continues to display "connecting"
>
> Test environment:
> - sip-communicator-1.0-alpha1-src.zip [also tried with nightly
> build of
> 6-June ]
> - apache-ant-1.7.0
> - jdk1.5.0_12 [ also tried the latest version 1.6]
> - jre1.5.0-12 [ also tried 1.6]
> - The sip-communicator clients run on Windows OS
> - The asterisk server runs on Debian Linux
>
> The screen Capture for ckient 1114 is attached
>
> We would appreciate your timely assistance
>
> Thanks,
> Gus
>
> _________________________________________________________________
> Get a preview of Live Earth, the hottest event this summer - only
> on MSN
> http://liveearth.msn.com?source=msntaglineliveearthhm
>
---------------------------------------------------------------------
> To unsubscribe, e-mail:
> dev-unsubscribe@sip-communicator.dev.java.net
> <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
> For additional commands, e-mail:
> dev-help@sip-communicator.dev.java.net
> <mailto:dev-help@sip-communicator.dev.java.net>
>

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#8

Hi,

test with last build of sip-communicator or version from CVS this problem is now corrected.

damencho

asmouta wrote:

···

Can you please send me your SIP configuration for Asterisk? I dont understand what's happening :frowning:

On 6/14/07, *Damian Minkov* < damencho@damencho.com > <mailto:damencho@damencho.com>> wrote:

    hi again,

    to me with asterisk 1.2... works ok - with no NAT between asterisk
    and
    sip-communicator.

    damencho

    asmouta wrote:
    > Can you please tell me wich Asterisk version are you using?? I'm
    > working with asterisk-1.2.18.
    >
    > On 6/9/07, *Gus Samba* < gus_samba@hotmail.com > <mailto:gus_samba@hotmail.com> > > <mailto:gus_samba@hotmail.com <mailto:gus_samba@hotmail.com>>> > wrote:
    >
    > Hi All,
    >
    > We are experiencing "one-way" audio problem between two
    > sip-communicator
    > clients.
    > Both clients register successfully with the asterisk server. The
    > clients and
    > server are all on the same LAN. There is no firewall, or
    router
    > in the test
    > configuration:
    >
    > When Client 1111 initiates a call to 1114:
    > - 1111 displays "alerting �" and generates a ringing tone
    > - 1114 displays "incoming.." and generates a ringing tone
    > When Client 1114 accepts the call:
    > - 1111 continues to ring for 2 secs, and displays
    "connected"
    > - 1114 status display changes from "incoming" to
    "connecting"
    > When 1111 speaks:
    > - 1114 clearly hears 1111
    > - But 1114 status continues to display "connecting"
    > When 1114 speaks:
    > - 1114 voice is echoed in 1114's earpiece
    > - 1111 does not hear 1114
    > - 1114 status continues to display "connecting"
    >
    > Test environment:
    > - sip-communicator-1.0-alpha1-src.zip [also tried with
    nightly
    > build of
    > 6-June ]
    > - apache-ant-1.7.0
    > - jdk1.5.0_12 [ also tried the latest version 1.6]
    > - jre1.5.0-12 [ also tried 1.6]
    > - The sip-communicator clients run on Windows OS
    > - The asterisk server runs on Debian Linux
    >
    > The screen Capture for ckient 1114 is attached
    >
    > We would appreciate your timely assistance
    >
    > Thanks,
    > Gus
    >
    > _________________________________________________________________
    > Get a preview of Live Earth, the hottest event this summer -
    only
    > on MSN
    > http://liveearth.msn.com?source=msntaglineliveearthhm
    >
    > ---------------------------------------------------------------------
    > To unsubscribe, e-mail:
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    <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
    > <mailto:dev-unsubscribe@sip-communicator.dev.java.net
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    > For additional commands, e-mail:
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    > <mailto:dev-help@sip-communicator.dev.java.net
    <mailto:dev-help@sip-communicator.dev.java.net>>
    >

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#9

Opps.... I can't! I'm working on a modified version, I mean I have changed
many things on the source version I download... :frowning: Maybe it will be helpful
if you tell me what should I modify to make it work too.. Hope that I'm not
annoying u much.. but it's so amazing since it was working a week ago and I
haven't modified my asterisk.. :frowning:

···

On 6/14/07, Damian Minkov <damencho@damencho.com> wrote:

Hi,

test with last build of sip-communicator or version from CVS this
problem is now corrected.

damencho

asmouta wrote:
> Can you please send me your SIP configuration for Asterisk? I dont
> understand what's happening :frowning:
>
> On 6/14/07, *Damian Minkov* < damencho@damencho.com > > <mailto:damencho@damencho.com>> wrote:
>
> hi again,
>
> to me with asterisk 1.2... works ok - with no NAT between asterisk
> and
> sip-communicator.
>
> damencho
>
> asmouta wrote:
> > Can you please tell me wich Asterisk version are you using?? I'm
> > working with asterisk-1.2.18.
> >
> > On 6/9/07, *Gus Samba* < gus_samba@hotmail.com > > <mailto:gus_samba@hotmail.com> > > > <mailto:gus_samba@hotmail.com <mailto:gus_samba@hotmail.com>>> > > wrote:
> >
> > Hi All,
> >
> > We are experiencing "one-way" audio problem between two
> > sip-communicator
> > clients.
> > Both clients register successfully with the asterisk server.
The
> > clients and
> > server are all on the same LAN. There is no firewall, or
> router
> > in the test
> > configuration:
> >
> > When Client 1111 initiates a call to 1114:
> > - 1111 displays "alerting …" and generates a ringing
tone
> > - 1114 displays "incoming.." and generates a ringing tone
> > When Client 1114 accepts the call:
> > - 1111 continues to ring for 2 secs, and displays
> "connected"
> > - 1114 status display changes from "incoming" to
> "connecting"
> > When 1111 speaks:
> > - 1114 clearly hears 1111
> > - But 1114 status continues to display "connecting"
> > When 1114 speaks:
> > - 1114 voice is echoed in 1114's earpiece
> > - 1111 does not hear 1114
> > - 1114 status continues to display "connecting"
> >
> > Test environment:
> > - sip-communicator-1.0-alpha1-src.zip [also tried with
> nightly
> > build of
> > 6-June ]
> > - apache-ant-1.7.0
> > - jdk1.5.0_12 [ also tried the latest version 1.6]
> > - jre1.5.0-12 [ also tried 1.6]
> > - The sip-communicator clients run on Windows OS
> > - The asterisk server runs on Debian Linux
> >
> > The screen Capture for ckient 1114 is attached
> >
> > We would appreciate your timely assistance
> >
> > Thanks,
> > Gus
> >
> _________________________________________________________________
> > Get a preview of Live Earth, the hottest event this summer -
> only
> > on MSN
> > http://liveearth.msn.com?source=msntaglineliveearthhm
> >
>
---------------------------------------------------------------------
> > To unsubscribe, e-mail:
> > dev-unsubscribe@sip-communicator.dev.java.net
> <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
> > <mailto:dev-unsubscribe@sip-communicator.dev.java.net
> <mailto:dev-unsubscribe@sip-communicator.dev.java.net>>
> > For additional commands, e-mail:
> > dev-help@sip-communicator.dev.java.net
> <mailto:dev-help@sip-communicator.dev.java.net>
> > <mailto:dev-help@sip-communicator.dev.java.net
> <mailto:dev-help@sip-communicator.dev.java.net>>
> >
>
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#10

Hi all,

since I've made the appropriate corrections to my SC version I got no more
this problem but those last days I've noticed another one. Now each time I
make a call from A to B, even if B is not connected or doesn't answer the
SC's A shows the "connected" status... What's happening??

Thanks a lot.

···

On 6/14/07, asmouta <asmouta@gmail.com> wrote:

Opps.... I can't! I'm working on a modified version, I mean I have changed
many things on the source version I download... :frowning: Maybe it will be helpful
if you tell me what should I modify to make it work too.. Hope that I'm not
annoying u much.. but it's so amazing since it was working a week ago and I
haven't modified my asterisk.. :frowning:

On 6/14/07, Damian Minkov <damencho@damencho.com> wrote:
>
> Hi,
>
> test with last build of sip-communicator or version from CVS this
> problem is now corrected.
>
> damencho
>
> asmouta wrote:
> > Can you please send me your SIP configuration for Asterisk? I dont
> > understand what's happening :frowning:
> >
> > On 6/14/07, *Damian Minkov* < damencho@damencho.com > > > <mailto:damencho@damencho.com >> wrote:
> >
> > hi again,
> >
> > to me with asterisk 1.2... works ok - with no NAT between asterisk
> > and
> > sip-communicator.
> >
> > damencho
> >
> > asmouta wrote:
> > > Can you please tell me wich Asterisk version are you using?? I'm
> > > working with asterisk-1.2.18.
> > >
> > > On 6/9/07, *Gus Samba* < gus_samba@hotmail.com > > > <mailto:gus_samba@hotmail.com> > > > > <mailto:gus_samba@hotmail.com <mailto: gus_samba@hotmail.com>>> > > > wrote:
> > >
> > > Hi All,
> > >
> > > We are experiencing "one-way" audio problem between two
> > > sip-communicator
> > > clients.
> > > Both clients register successfully with the asterisk server.
> The
> > > clients and
> > > server are all on the same LAN. There is no firewall, or
> > router
> > > in the test
> > > configuration:
> > >
> > > When Client 1111 initiates a call to 1114:
> > > - 1111 displays "alerting …" and generates a ringing
> tone
> > > - 1114 displays "incoming.." and generates a ringing
> tone
> > > When Client 1114 accepts the call:
> > > - 1111 continues to ring for 2 secs, and displays
> > "connected"
> > > - 1114 status display changes from "incoming" to
> > "connecting"
> > > When 1111 speaks:
> > > - 1114 clearly hears 1111
> > > - But 1114 status continues to display "connecting"
> > > When 1114 speaks:
> > > - 1114 voice is echoed in 1114's earpiece
> > > - 1111 does not hear 1114
> > > - 1114 status continues to display "connecting"
> > >
> > > Test environment:
> > > - sip-communicator-1.0-alpha1-src.zip [also tried with
> > nightly
> > > build of
> > > 6-June ]
> > > - apache-ant-1.7.0
> > > - jdk1.5.0_12 [ also tried the latest version 1.6]
> > > - jre1.5.0-12 [ also tried 1.6]
> > > - The sip-communicator clients run on Windows OS
> > > - The asterisk server runs on Debian Linux
> > >
> > > The screen Capture for ckient 1114 is attached
> > >
> > > We would appreciate your timely assistance
> > >
> > > Thanks,
> > > Gus
> > >
> > _________________________________________________________________
> > > Get a preview of Live Earth, the hottest event this summer -
> > only
> > > on MSN
> > > http://liveearth.msn.com?source=msntaglineliveearthhm
> > >
> >
> ---------------------------------------------------------------------
> > > To unsubscribe, e-mail:
> > > dev-unsubscribe@sip-communicator.dev.java.net
> > <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
> > > <mailto: dev-unsubscribe@sip-communicator.dev.java.net
> > <mailto:dev-unsubscribe@sip-communicator.dev.java.net >>
> > > For additional commands, e-mail:
> > > dev-help@sip-communicator.dev.java.net
> > <mailto: dev-help@sip-communicator.dev.java.net>
> > > <mailto:dev-help@sip-communicator.dev.java.net
> > <mailto: dev-help@sip-communicator.dev.java.net>>
> > >
> >
> ---------------------------------------------------------------------
> > To unsubscribe, e-mail:
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> > <mailto:dev-unsubscribe@sip-communicator.dev.java.net >
> > For additional commands, e-mail:
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>
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> For additional commands, e-mail: dev-help@sip-communicator.dev.java.net
>


#11

hi,

sounds like asterisk is answering the call. It can answer the caller and then dial the other party
exten => s,1,Answer
exten => s,2,Dial...
.......

Check this. and check the dump.

damencho

asmouta wrote:

···

Hi all,

since I've made the appropriate corrections to my SC version I got no more this problem but those last days I've noticed another one. Now each time I make a call from A to B, even if B is not connected or doesn't answer the SC's A shows the "connected" status... What's happening??

Thanks a lot.

On 6/14/07, *asmouta* <asmouta@gmail.com <mailto:asmouta@gmail.com>> > wrote:

    Opps.... I can't! I'm working on a modified version, I mean I have
    changed many things on the source version I download... :frowning: Maybe
    it will be helpful if you tell me what should I modify to make it
    work too.. Hope that I'm not annoying u much.. but it's so amazing
    since it was working a week ago and I haven't modified my
    asterisk.. :frowning:

    On 6/14/07, *Damian Minkov* < damencho@damencho.com > <mailto:damencho@damencho.com>> wrote:

        Hi,

        test with last build of sip-communicator or version from CVS this
        problem is now corrected.

        damencho

        asmouta wrote:
        > Can you please send me your SIP configuration for Asterisk? I
        dont
        > understand what's happening :frowning:
        >
        > On 6/14/07, *Damian Minkov* < damencho@damencho.com > <mailto:damencho@damencho.com> > > <mailto:damencho@damencho.com > <mailto:damencho@damencho.com>>> wrote:
        >
        > hi again,
        >
        > to me with asterisk 1.2... works ok - with no NAT between
        asterisk
        > and
        > sip-communicator.
        >
        > damencho
        >
        > asmouta wrote:
        > > Can you please tell me wich Asterisk version are you
        using?? I'm
        > > working with asterisk-1.2.18.
        > >
        > > On 6/9/07, *Gus Samba* < gus_samba@hotmail.com
        <mailto:gus_samba@hotmail.com>
        > <mailto:gus_samba@hotmail.com <mailto:gus_samba@hotmail.com>>
        > > <mailto: gus_samba@hotmail.com
        <mailto:gus_samba@hotmail.com> <mailto: gus_samba@hotmail.com
        <mailto:gus_samba@hotmail.com>>>>
        > wrote:
        > >
        > > Hi All,
        > >
        > > We are experiencing "one-way" audio problem between
        two
        > > sip-communicator
        > > clients.
        > > Both clients register successfully with the
        asterisk server. The
        > > clients and
        > > server are all on the same LAN. There is no
        firewall, or
        > router
        > > in the test
        > > configuration:
        > >
        > > When Client 1111 initiates a call to 1114:
        > > - 1111 displays "alerting �" and generates a
        ringing tone
        > > - 1114 displays "incoming.." and generates a
        ringing tone
        > > When Client 1114 accepts the call:
        > > - 1111 continues to ring for 2 secs, and displays
        > "connected"
        > > - 1114 status display changes from "incoming" to
        > "connecting"
        > > When 1111 speaks:
        > > - 1114 clearly hears 1111
        > > - But 1114 status continues to display
        "connecting"
        > > When 1114 speaks:
        > > - 1114 voice is echoed in 1114's earpiece
        > > - 1111 does not hear 1114
        > > - 1114 status continues to display "connecting"
        > >
        > > Test environment:
        > > - sip-communicator-1.0-alpha1-src.zip [also tried
        with
        > nightly
        > > build of
        > > 6-June ]
        > > - apache-ant-1.7.0
        > > - jdk1.5.0_12 [ also tried the latest version 1.6]
        > > - jre1.5.0-12 [ also tried 1.6]
        > > - The sip-communicator clients run on Windows OS
        > > - The asterisk server runs on Debian Linux
        > >
        > > The screen Capture for ckient 1114 is attached
        > >
        > > We would appreciate your timely assistance
        > >
        > > Thanks,
        > > Gus
        > >
        > _________________________________________________________________
        > > Get a preview of Live Earth, the hottest event this
        summer -
        > only
        > > on MSN
        > > http://liveearth.msn.com?source=msntaglineliveearthhm
        <http://liveearth.msn.com?source=msntaglineliveearthhm>
        > >
        > ---------------------------------------------------------------------
        > > To unsubscribe, e-mail:
        > > dev-unsubscribe@sip-communicator.dev.java.net
        <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
        > <mailto:dev-unsubscribe@sip-communicator.dev.java.net
        <mailto:dev-unsubscribe@sip-communicator.dev.java.net>>
        > > <mailto:
        dev-unsubscribe@sip-communicator.dev.java.net
        <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
        > <mailto: dev-unsubscribe@sip-communicator.dev.java.net
        <mailto:dev-unsubscribe@sip-communicator.dev.java.net>>>
        > > For additional commands, e-mail:
        > > dev-help@sip-communicator.dev.java.net
        <mailto:dev-help@sip-communicator.dev.java.net>
        > <mailto: dev-help@sip-communicator.dev.java.net
        <mailto:dev-help@sip-communicator.dev.java.net>>
        > > <mailto:dev-help@sip-communicator.dev.java.net
        <mailto:dev-help@sip-communicator.dev.java.net>
        > <mailto: dev-help@sip-communicator.dev.java.net
        <mailto:dev-help@sip-communicator.dev.java.net>>>
        > >
        >
        > ---------------------------------------------------------------------
        > To unsubscribe, e-mail:
        > dev-unsubscribe@sip-communicator.dev.java.net
        <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
        > <mailto: dev-unsubscribe@sip-communicator.dev.java.net
        <mailto:dev-unsubscribe@sip-communicator.dev.java.net>>
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        <mailto:dev-help@sip-communicator.dev.java.net>
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        >

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#12

Hi Damian,

here is my Astersik configuration :

exten => _15.,1,Dial(SIP/${EXTEN},10,rt)

So actually I dont think that's related to Astersik. Thanks for your answer.

···

On 6/25/07, Damian Minkov <damencho@damencho.com> wrote:

hi,

sounds like asterisk is answering the call. It can answer the caller and
then dial the other party
exten => s,1,Answer
exten => s,2,Dial...
.......

Check this. and check the dump.

damencho

asmouta wrote:
> Hi all,
>
> since I've made the appropriate corrections to my SC version I got no
> more this problem but those last days I've noticed another one. Now
> each time I make a call from A to B, even if B is not connected or
> doesn't answer the SC's A shows the "connected" status... What's
> happening??
>
> Thanks a lot.
>
> On 6/14/07, *asmouta* <asmouta@gmail.com <mailto:asmouta@gmail.com>> > > wrote:
>
> Opps.... I can't! I'm working on a modified version, I mean I have
> changed many things on the source version I download... :frowning: Maybe
> it will be helpful if you tell me what should I modify to make it
> work too.. Hope that I'm not annoying u much.. but it's so amazing
> since it was working a week ago and I haven't modified my
> asterisk.. :frowning:
>
> On 6/14/07, *Damian Minkov* < damencho@damencho.com > > <mailto:damencho@damencho.com>> wrote:
>
> Hi,
>
> test with last build of sip-communicator or version from CVS
this
> problem is now corrected.
>
> damencho
>
> asmouta wrote:
> > Can you please send me your SIP configuration for Asterisk? I
> dont
> > understand what's happening :frowning:
> >
> > On 6/14/07, *Damian Minkov* < damencho@damencho.com > > <mailto:damencho@damencho.com> > > > <mailto:damencho@damencho.com > > <mailto:damencho@damencho.com>>> wrote:
> >
> > hi again,
> >
> > to me with asterisk 1.2... works ok - with no NAT between
> asterisk
> > and
> > sip-communicator.
> >
> > damencho
> >
> > asmouta wrote:
> > > Can you please tell me wich Asterisk version are you
> using?? I'm
> > > working with asterisk-1.2.18.
> > >
> > > On 6/9/07, *Gus Samba* < gus_samba@hotmail.com
> <mailto:gus_samba@hotmail.com>
> > <mailto:gus_samba@hotmail.com <mailto:
gus_samba@hotmail.com>>
> > > <mailto: gus_samba@hotmail.com
> <mailto:gus_samba@hotmail.com> <mailto: gus_samba@hotmail.com
> <mailto:gus_samba@hotmail.com>>>>
> > wrote:
> > >
> > > Hi All,
> > >
> > > We are experiencing "one-way" audio problem between
> two
> > > sip-communicator
> > > clients.
> > > Both clients register successfully with the
> asterisk server. The
> > > clients and
> > > server are all on the same LAN. There is no
> firewall, or
> > router
> > > in the test
> > > configuration:
> > >
> > > When Client 1111 initiates a call to 1114:
> > > - 1111 displays "alerting …" and generates a
> ringing tone
> > > - 1114 displays "incoming.." and generates a
> ringing tone
> > > When Client 1114 accepts the call:
> > > - 1111 continues to ring for 2 secs, and
displays
> > "connected"
> > > - 1114 status display changes from "incoming"
to
> > "connecting"
> > > When 1111 speaks:
> > > - 1114 clearly hears 1111
> > > - But 1114 status continues to display
> "connecting"
> > > When 1114 speaks:
> > > - 1114 voice is echoed in 1114's earpiece
> > > - 1111 does not hear 1114
> > > - 1114 status continues to display "connecting"
> > >
> > > Test environment:
> > > - sip-communicator-1.0-alpha1-src.zip [also tried
> with
> > nightly
> > > build of
> > > 6-June ]
> > > - apache-ant-1.7.0
> > > - jdk1.5.0_12 [ also tried the latest version 1.6
]
> > > - jre1.5.0-12 [ also tried 1.6]
> > > - The sip-communicator clients run on Windows OS
> > > - The asterisk server runs on Debian Linux
> > >
> > > The screen Capture for ckient 1114 is attached
> > >
> > > We would appreciate your timely assistance
> > >
> > > Thanks,
> > > Gus
> > >
> >
>
_________________________________________________________________
> > > Get a preview of Live Earth, the hottest event this
> summer -
> > only
> > > on MSN
> > >
> http://liveearth.msn.com?source=msntaglineliveearthhm
> <http://liveearth.msn.com?source=msntaglineliveearthhm>
> > >
> >
>
---------------------------------------------------------------------
> > > To unsubscribe, e-mail:
> > > dev-unsubscribe@sip-communicator.dev.java.net
> <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
> > <mailto:dev-unsubscribe@sip-communicator.dev.java.net
> <mailto:dev-unsubscribe@sip-communicator.dev.java.net>>
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> <mailto:dev-unsubscribe@sip-communicator.dev.java.net>>>
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> > > dev-help@sip-communicator.dev.java.net
> <mailto:dev-help@sip-communicator.dev.java.net>
> > <mailto: dev-help@sip-communicator.dev.java.net
> <mailto:dev-help@sip-communicator.dev.java.net>>
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> <mailto:dev-help@sip-communicator.dev.java.net>
> > <mailto: dev-help@sip-communicator.dev.java.net
> <mailto:dev-help@sip-communicator.dev.java.net>>>
> > >
> >
>
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#13

Hi!

I'm starting to think that the problem is the unsorted processing of
h263 packets. I'd like to know if is there any component that controls
this and sorts the packets.

Thanks. Pablo.

···

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#14

Hi Pablo,

You might want to ask this on the JMF mailing list.

Emil

PABLO LOPEZ GARCIA wrote:

···

Hi!

I'm starting to think that the problem is the unsorted processing of h263 packets. I'd like to know if is there any component that controls this and sorts the packets.

Thanks. Pablo.

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