[sip-comm-dev] How to handle incoming calls?


#1

Hi,
I just switched from Qutecomm to SIP Communicator. SIP Communicator looks
more stable, even with the current alpha version. Great work!

But.... (there is always a but)

Everything works fine, chatting works fine, chat notifcation works fine, SIP
calling works fine.Only, when I receive an incoming call, nothing happens.
No sound, no pop up. I have checked the setting is tool -> notifcations.
Both are activated.

When I look in the call history, only outgoing calls are logged...

What am I missing? Please let me know, I want to fully switch to SIP
Communicator.

Thanks,
Eelco Mulder


#2

Hi,
Just to be sure. Has this message arrived the correct queue?

Second, other people can receive inbound sip calls without any problems? How
does this work? (I am using Voipcheap mementarily, and inbound calls with
Qutecomm did work flawless).

Thanks for your help in advance!

Regards,
Eelco Mulder

···

2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com>

Hi,
I just switched from Qutecomm to SIP Communicator. SIP Communicator looks
more stable, even with the current alpha version. Great work!

But.... (there is always a but)

Everything works fine, chatting works fine, chat notifcation works fine,
SIP calling works fine.Only, when I receive an incoming call, nothing
happens. No sound, no pop up. I have checked the setting is tool ->
notifcations. Both are activated.

When I look in the call history, only outgoing calls are logged...

What am I missing? Please let me know, I want to fully switch to SIP
Communicator.

Thanks,
Eelco Mulder


#3

Hey Eelco,

Eelco Mulder wrote:

Hi,

Just to be sure. Has this message arrived the correct queue?

Yes.

Second, other people can receive inbound sip calls without any problems?

Yes.

How does this work? (I am using Voipcheap mementarily, and inbound calls
with Qutecomm did work flawless).

You may want to get a wireshark dump and see how the incoming INVITEs
look. You can send the dump or a link to it here in case you are not
able to figure it out by yourself.

It would also help to have a look at your log files.

Cheers,
Emil

···

Thanks for your help in advance!

Regards,
Eelco Mulder

2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
<mailto:eelco_mulder@hotmail.com>>

    Hi,

    I just switched from Qutecomm to SIP Communicator. SIP Communicator
    looks more stable, even with the current alpha version. Great work!

    But.... (there is always a but)

    Everything works fine, chatting works fine, chat notifcation works
    fine, SIP calling works fine.Only, when I receive an incoming call,
    nothing happens. No sound, no pop up. I have checked the setting is
    tool -> notifcations. Both are activated.

    When I look in the call history, only outgoing calls are logged...

    What am I missing? Please let me know, I want to fully switch to SIP
    Communicator.

    Thanks,
    Eelco Mulder

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#4

Hi Emil,
Thanks for your prompt reply!

I'd love to check the logfiles or wireshark dump, but please have some
patience with me. I am new to sip comm, so I don't know where to find these.

I have checked the eventviewer logs, but there are no entries in here when I
call my SIP number.

I have checked the Program File/ Sip Communicator directory, but did not
find any logging or entry of today.

Can you please link me to where and how I can retrieve the data?

Thanks!
Eelco Mulder

···

2009/9/18 Emil Ivov <emcho@sip-communicator.org>

Hey Eelco,

Eelco Mulder wrote:
> Hi,
>
> Just to be sure. Has this message arrived the correct queue?

Yes.

> Second, other people can receive inbound sip calls without any problems?

Yes.

> How does this work? (I am using Voipcheap mementarily, and inbound calls
> with Qutecomm did work flawless).

You may want to get a wireshark dump and see how the incoming INVITEs
look. You can send the dump or a link to it here in case you are not
able to figure it out by yourself.

It would also help to have a look at your log files.

Cheers,
Emil

>
> Thanks for your help in advance!
>
> Regards,
> Eelco Mulder
>
> 2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
> <mailto:eelco_mulder@hotmail.com>>
>
> Hi,
>
> I just switched from Qutecomm to SIP Communicator. SIP Communicator
> looks more stable, even with the current alpha version. Great work!
>
> But.... (there is always a but)
>
> Everything works fine, chatting works fine, chat notifcation works
> fine, SIP calling works fine.Only, when I receive an incoming call,
> nothing happens. No sound, no pop up. I have checked the setting is
> tool -> notifcations. Both are activated.
>
> When I look in the call history, only outgoing calls are logged...
>
> What am I missing? Please let me know, I want to fully switch to SIP
> Communicator.
>
> Thanks,
> Eelco Mulder

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#5

Hi Eelco,

We just created a FAQ entry at http://www.sip-communicator.org/index.php/Documentation/FAQ#logs on the subject of the location of the log files. Please refer to it.

Regards,
Lubomir

···

On 18.09.2009 11:02, Eelco Mulder wrote:

Hi Emil,

Thanks for your prompt reply!

I'd love to check the logfiles or wireshark dump, but please have some
patience with me. I am new to sip comm, so I don't know where to find these.

I have checked the eventviewer logs, but there are no entries in here
when I call my SIP number.

I have checked the Program File/ Sip Communicator directory, but did not
find any logging or entry of today.

Can you please link me to where and how I can retrieve the data?

Thanks!
Eelco Mulder

2009/9/18 Emil Ivov <emcho@sip-communicator.org
<mailto:emcho@sip-communicator.org>>

    Hey Eelco,

    Eelco Mulder wrote:
     > Hi,
     >
     > Just to be sure. Has this message arrived the correct queue?

    Yes.

     > Second, other people can receive inbound sip calls without any
    problems?

    Yes.

     > How does this work? (I am using Voipcheap mementarily, and
    inbound calls
     > with Qutecomm did work flawless).

    You may want to get a wireshark dump and see how the incoming INVITEs
    look. You can send the dump or a link to it here in case you are not
    able to figure it out by yourself.

    It would also help to have a look at your log files.

    Cheers,
    Emil

     >
     > Thanks for your help in advance!
     >
     > Regards,
     > Eelco Mulder
     >
     > 2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
     > <mailto:eelco_mulder@hotmail.com <mailto:eelco_mulder@hotmail.com>>>
     >
     > Hi,
     >
     > I just switched from Qutecomm to SIP Communicator. SIP
    Communicator
     > looks more stable, even with the current alpha version. Great
    work!
     >
     > But.... (there is always a but)
     >
     > Everything works fine, chatting works fine, chat notifcation
    works
     > fine, SIP calling works fine.Only, when I receive an incoming
    call,
     > nothing happens. No sound, no pop up. I have checked the
    setting is
     > tool -> notifcations. Both are activated.
     >
     > When I look in the call history, only outgoing calls are
    logged...
     >
     > What am I missing? Please let me know, I want to fully switch
    to SIP
     > Communicator.
     >
     > Thanks,
     > Eelco Mulder

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#6

Great, thanks!
I have been digging in the logs and I think I found the incoming call entry
in the logs. Please let me know what I can read in it or what else I can
check.... fyi, when I call my SIP number, I hear my pstn phone ringing, but
SIP comm does not give me a popup or sound, nor can I pick up the phone
somehow.

Thanks,
Eelco Mulder

11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug output
from the JAIN-SIP stack: semRelease
]]]]gov.nist.javax.sip.stack.SIPServerTransaction@be009483
11:50:33.903 FINER: impl.protocol.sip.SipLogger.logStackTrace().41 JAIN-SIP
stack trace
java.lang.Throwable
at
net.java.sip.communicator.impl.protocol.sip.SipLogger.logStackTrace(SipLogger.java:41)
at
gov.nist.javax.sip.stack.SIPTransaction.semRelease(SIPTransaction.java:1198)
at
gov.nist.javax.sip.stack.SIPTransaction.releaseSem(SIPTransaction.java:1184)
at
gov.nist.javax.sip.stack.SIPServerTransaction.releaseSem(SIPServerTransaction.java:1645)
at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:254)
at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
at java.lang.Thread.run(Unknown Source)
11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug output
from the JAIN-SIP stack: removePendingTx:
z9hg4bk9057a6213f604d23909211763ef079b1
11:50:35.653 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug output
from the JAIN-SIP stack: UDPMessageChannel: processIncomingDataPacket :
peerAddress = 194.120.0.198/5060 Length = 537
11:50:35.653 FINE: impl.protocol.sip.AddressResolverImpl.resolveAddress().96
Returning hop: 123.456.0.999:5060/UDP
11:50:35.653 FINER: impl.protocol.sip.SipLogger.logMessage().188 JAIN-SIP
received message from "123.456.0.999:5060" to "0.0.0.0:5060" at
1253267435653
CANCEL sip:oclee_redlum@10.31.119.53:5060;transport=udp;registering_acc=voipcheap_com
SIP/2.0
Via: SIP/2.0/UDP
123.456.0.999:5060;branch=z9hL4bK9057a6213f604d23909241763ef079b1

From: <sip:0031234128888@voipcheap.com:5060
;tag=c11710acc12b10ac4aa6315a20dbd4

Contact: <sip:0031243128888@194.120.0.198:5060>
Call-ID: 2649a604f40b482a8c011393187bc6ac
CSeq: 1 CANCEL
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

*(please note I have altered IP addresses and telephonenumbers for privacy
reasons)*

···

To: <sip:oclee_redlum@10.31.119.53:5060>

2009/9/18 Lubomir Marinov <lubomir.marinov@gmail.com>

Hi Eelco,

We just created a FAQ entry at
http://www.sip-communicator.org/index.php/Documentation/FAQ#logs on the
subject of the location of the log files. Please refer to it.

Regards,
Lubomir

On 18.09.2009 11:02, Eelco Mulder wrote:

Hi Emil,

Thanks for your prompt reply!

I'd love to check the logfiles or wireshark dump, but please have some
patience with me. I am new to sip comm, so I don't know where to find
these.

I have checked the eventviewer logs, but there are no entries in here
when I call my SIP number.

I have checked the Program File/ Sip Communicator directory, but did not
find any logging or entry of today.

Can you please link me to where and how I can retrieve the data?

Thanks!
Eelco Mulder

2009/9/18 Emil Ivov <emcho@sip-communicator.org
<mailto:emcho@sip-communicator.org>>

   Hey Eelco,

   Eelco Mulder wrote:
    > Hi,
    >
    > Just to be sure. Has this message arrived the correct queue?

   Yes.

    > Second, other people can receive inbound sip calls without any
   problems?

   Yes.

    > How does this work? (I am using Voipcheap mementarily, and
   inbound calls
    > with Qutecomm did work flawless).

   You may want to get a wireshark dump and see how the incoming INVITEs
   look. You can send the dump or a link to it here in case you are not
   able to figure it out by yourself.

   It would also help to have a look at your log files.

   Cheers,
   Emil

    >
    > Thanks for your help in advance!
    >
    > Regards,
    > Eelco Mulder
    >
    > 2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
   <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com <mailto:eelco_mulder@hotmail.com
>>>
    >
    > Hi,
    >
    > I just switched from Qutecomm to SIP Communicator. SIP
   Communicator
    > looks more stable, even with the current alpha version. Great
   work!
    >
    > But.... (there is always a but)
    >
    > Everything works fine, chatting works fine, chat notifcation
   works
    > fine, SIP calling works fine.Only, when I receive an incoming
   call,
    > nothing happens. No sound, no pop up. I have checked the
   setting is
    > tool -> notifcations. Both are activated.
    >
    > When I look in the call history, only outgoing calls are
   logged...
    >
    > What am I missing? Please let me know, I want to fully switch
   to SIP
    > Communicator.
    >
    > Thanks,
    > Eelco Mulder

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#7

Eelco,

I only see a CANCEL request in here and this doesn't really tell me that
much. We would need your complete log file in order to be able to help.

Emil

Eelco Mulder wrote:

···

Great, thanks!

I have been digging in the logs and I think I found the incoming call
entry in the logs. Please let me know what I can read in it or what else
I can check.... fyi, when I call my SIP number, I hear my pstn phone
ringing, but SIP comm does not give me a popup or sound, nor can I pick
up the phone somehow.

Thanks,
Eelco Mulder

11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
output from the JAIN-SIP stack: semRelease
]]]]gov.nist.javax.sip.stack.SIPServerTransaction@be009483
11:50:33.903 FINER: impl.protocol.sip.SipLogger.logStackTrace().41
JAIN-SIP stack trace
java.lang.Throwable
at
net.java.sip.communicator.impl.protocol.sip.SipLogger.logStackTrace(SipLogger.java:41)
at
gov.nist.javax.sip.stack.SIPTransaction.semRelease(SIPTransaction.java:1198)
at
gov.nist.javax.sip.stack.SIPTransaction.releaseSem(SIPTransaction.java:1184)
at
gov.nist.javax.sip.stack.SIPServerTransaction.releaseSem(SIPServerTransaction.java:1645)
at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:254)
at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
at java.lang.Thread.run(Unknown Source)
11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
output from the JAIN-SIP stack: removePendingTx:
z9hg4bk9057a6213f604d23909211763ef079b1
11:50:35.653 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
output from the JAIN-SIP stack: UDPMessageChannel:
processIncomingDataPacket : peerAddress = 194.120.0.198/5060
<http://194.120.0.198/5060> Length = 537
11:50:35.653 FINE:
impl.protocol.sip.AddressResolverImpl.resolveAddress().96 Returning hop:
123.456.0.999:5060/UDP
11:50:35.653 FINER: impl.protocol.sip.SipLogger.logMessage().188
JAIN-SIP received message from "123.456.0.999:5060" to "0.0.0.0:5060
<http://0.0.0.0:5060>" at 1253267435653
CANCEL
sip:oclee_redlum@10.31.119.53:5060;transport=udp;registering_acc=voipcheap_com
SIP/2.0
Via: SIP/2.0/UDP
123.456.0.999:5060;branch=z9hL4bK9057a6213f604d23909241763ef079b1
From: <sip:0031234128888@voipcheap.com:5060
<http://sip:0031234128888@voipcheap.com:5060>>;tag=c11710acc12b10ac4aa6315a20dbd4
To: <sip:oclee_redlum@10.31.119.53:5060
<http://sip:oclee_redlum@10.31.119.53:5060>>
Contact: <sip:0031243128888@194.120.0.198:5060
<http://sip:0031243128888@194.120.0.198:5060>>
Call-ID: 2649a604f40b482a8c011393187bc6ac
CSeq: 1 CANCEL
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

/(please note I have altered IP addresses and telephonenumbers for
privacy reasons)/

2009/9/18 Lubomir Marinov <lubomir.marinov@gmail.com
<mailto:lubomir.marinov@gmail.com>>

    Hi Eelco,

    We just created a FAQ entry at
    http://www.sip-communicator.org/index.php/Documentation/FAQ#logs on
    the subject of the location of the log files. Please refer to it.

    Regards,
    Lubomir

    On 18.09.2009 11:02, Eelco Mulder wrote:

        Hi Emil,

        Thanks for your prompt reply!

        I'd love to check the logfiles or wireshark dump, but please
        have some
        patience with me. I am new to sip comm, so I don't know where to
        find these.

        I have checked the eventviewer logs, but there are no entries in
        here
        when I call my SIP number.

        I have checked the Program File/ Sip Communicator directory, but
        did not
        find any logging or entry of today.

        Can you please link me to where and how I can retrieve the data?

        Thanks!
        Eelco Mulder

        2009/9/18 Emil Ivov <emcho@sip-communicator.org
        <mailto:emcho@sip-communicator.org>
        <mailto:emcho@sip-communicator.org
        <mailto:emcho@sip-communicator.org>>>

           Hey Eelco,

           Eelco Mulder wrote:
            > Hi,
            >
            > Just to be sure. Has this message arrived the correct queue?

           Yes.

            > Second, other people can receive inbound sip calls without any
           problems?

           Yes.

            > How does this work? (I am using Voipcheap mementarily, and
           inbound calls
            > with Qutecomm did work flawless).

           You may want to get a wireshark dump and see how the incoming
        INVITEs
           look. You can send the dump or a link to it here in case you
        are not
           able to figure it out by yourself.

           It would also help to have a look at your log files.

           Cheers,
           Emil

            >
            > Thanks for your help in advance!
            >
            > Regards,
            > Eelco Mulder
            >
            > 2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
        <mailto:eelco_mulder@hotmail.com>
           <mailto:eelco_mulder@hotmail.com
        <mailto:eelco_mulder@hotmail.com>>
            > <mailto:eelco_mulder@hotmail.com
        <mailto:eelco_mulder@hotmail.com>
        <mailto:eelco_mulder@hotmail.com
        <mailto:eelco_mulder@hotmail.com>>>>

            >
            > Hi,
            >
            > I just switched from Qutecomm to SIP Communicator. SIP
           Communicator
            > looks more stable, even with the current alpha
        version. Great
           work!
            >
            > But.... (there is always a but)
            >
            > Everything works fine, chatting works fine, chat
        notifcation
           works
            > fine, SIP calling works fine.Only, when I receive an
        incoming
           call,
            > nothing happens. No sound, no pop up. I have checked the
           setting is
            > tool -> notifcations. Both are activated.
            >
            > When I look in the call history, only outgoing calls are
           logged...
            >
            > What am I missing? Please let me know, I want to fully
        switch
           to SIP
            > Communicator.
            >
            > Thanks,
            > Eelco Mulder

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    <mailto:dev-unsubscribe@sip-communicator.dev.java.net>
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--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#8

No problem, but given all the personal data, I would like to only send it to
a private email address, not a group address...
Which one can I use?

Thanks,
Eelco Mulder

···

2009/9/18 Emil Ivov <emcho@sip-communicator.org>

Eelco,

I only see a CANCEL request in here and this doesn't really tell me that
much. We would need your complete log file in order to be able to help.

Emil

Eelco Mulder wrote:
> Great, thanks!
>
> I have been digging in the logs and I think I found the incoming call
> entry in the logs. Please let me know what I can read in it or what else
> I can check.... fyi, when I call my SIP number, I hear my pstn phone
> ringing, but SIP comm does not give me a popup or sound, nor can I pick
> up the phone somehow.
>
> Thanks,
> Eelco Mulder
>
> 11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
> output from the JAIN-SIP stack: semRelease
> ]]]]gov.nist.javax.sip.stack.SIPServerTransaction@be009483
> 11:50:33.903 FINER: impl.protocol.sip.SipLogger.logStackTrace().41
> JAIN-SIP stack trace
> java.lang.Throwable
> at
>
net.java.sip.communicator.impl.protocol.sip.SipLogger.logStackTrace(SipLogger.java:41)
> at
>
gov.nist.javax.sip.stack.SIPTransaction.semRelease(SIPTransaction.java:1198)
> at
>
gov.nist.javax.sip.stack.SIPTransaction.releaseSem(SIPTransaction.java:1184)
> at
>
gov.nist.javax.sip.stack.SIPServerTransaction.releaseSem(SIPServerTransaction.java:1645)
> at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:254)
> at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
> at java.lang.Thread.run(Unknown Source)
> 11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
> output from the JAIN-SIP stack: removePendingTx:
> z9hg4bk9057a6213f604d23909211763ef079b1
> 11:50:35.653 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
> output from the JAIN-SIP stack: UDPMessageChannel:
> processIncomingDataPacket : peerAddress = 194.120.0.198/5060
> <http://194.120.0.198/5060> Length = 537
> 11:50:35.653 FINE:
> impl.protocol.sip.AddressResolverImpl.resolveAddress().96 Returning hop:
> 123.456.0.999:5060/UDP
> 11:50:35.653 FINER: impl.protocol.sip.SipLogger.logMessage().188
> JAIN-SIP received message from "123.456.0.999:5060" to "0.0.0.0:5060
> <http://0.0.0.0:5060>" at 1253267435653
> CANCEL
> sip:oclee_redlum@10.31.119.53:5060
;transport=udp;registering_acc=voipcheap_com
> SIP/2.0
> Via: SIP/2.0/UDP
> 123.456.0.999:5060;branch=z9hL4bK9057a6213f604d23909241763ef079b1
> From: <sip:0031234128888@voipcheap.com:5060
> <http://sip:0031234128888@voipcheap.com:5060
>>;tag=c11710acc12b10ac4aa6315a20dbd4
> To: <sip:oclee_redlum@10.31.119.53:5060
> <http://sip:oclee_redlum@10.31.119.53:5060>>
> Contact: <sip:0031243128888@194.120.0.198:5060
> <http://sip:0031243128888@194.120.0.198:5060>>
> Call-ID: 2649a604f40b482a8c011393187bc6ac
> CSeq: 1 CANCEL
> Server: (Very nice Sip Registrar/Proxy Server)
> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
> Content-Length: 0
>
> /(please note I have altered IP addresses and telephonenumbers for
> privacy reasons)/
>
>
> 2009/9/18 Lubomir Marinov <lubomir.marinov@gmail.com
> <mailto:lubomir.marinov@gmail.com>>
>
> Hi Eelco,
>
> We just created a FAQ entry at
> http://www.sip-communicator.org/index.php/Documentation/FAQ#logs on
> the subject of the location of the log files. Please refer to it.
>
> Regards,
> Lubomir
>
>
> On 18.09.2009 11:02, Eelco Mulder wrote:
>
> Hi Emil,
>
> Thanks for your prompt reply!
>
> I'd love to check the logfiles or wireshark dump, but please
> have some
> patience with me. I am new to sip comm, so I don't know where to
> find these.
>
> I have checked the eventviewer logs, but there are no entries in
> here
> when I call my SIP number.
>
> I have checked the Program File/ Sip Communicator directory, but
> did not
> find any logging or entry of today.
>
> Can you please link me to where and how I can retrieve the data?
>
> Thanks!
> Eelco Mulder
>
> 2009/9/18 Emil Ivov <emcho@sip-communicator.org
> <mailto:emcho@sip-communicator.org>
> <mailto:emcho@sip-communicator.org
> <mailto:emcho@sip-communicator.org>>>
>
>
> Hey Eelco,
>
> Eelco Mulder wrote:
> > Hi,
> >
> > Just to be sure. Has this message arrived the correct
queue?
>
> Yes.
>
> > Second, other people can receive inbound sip calls without
any
> problems?
>
> Yes.
>
> > How does this work? (I am using Voipcheap mementarily, and
> inbound calls
> > with Qutecomm did work flawless).
>
> You may want to get a wireshark dump and see how the incoming
> INVITEs
> look. You can send the dump or a link to it here in case you
> are not
> able to figure it out by yourself.
>
> It would also help to have a look at your log files.
>
> Cheers,
> Emil
>
> >
> > Thanks for your help in advance!
> >
> > Regards,
> > Eelco Mulder
> >
> > 2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
> <mailto:eelco_mulder@hotmail.com>
> <mailto:eelco_mulder@hotmail.com
> <mailto:eelco_mulder@hotmail.com>>
> > <mailto:eelco_mulder@hotmail.com
> <mailto:eelco_mulder@hotmail.com>
> <mailto:eelco_mulder@hotmail.com
> <mailto:eelco_mulder@hotmail.com>>>>
>
> >
> > Hi,
> >
> > I just switched from Qutecomm to SIP Communicator. SIP
> Communicator
> > looks more stable, even with the current alpha
> version. Great
> work!
> >
> > But.... (there is always a but)
> >
> > Everything works fine, chatting works fine, chat
> notifcation
> works
> > fine, SIP calling works fine.Only, when I receive an
> incoming
> call,
> > nothing happens. No sound, no pop up. I have checked
the
> setting is
> > tool -> notifcations. Both are activated.
> >
> > When I look in the call history, only outgoing calls
are
> logged...
> >
> > What am I missing? Please let me know, I want to fully
> switch
> to SIP
> > Communicator.
> >
> > Thanks,
> > Eelco Mulder
>
>
> ---------------------------------------------------------------------
> 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>
>
>

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

---------------------------------------------------------------------
To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
For additional commands, e-mail: dev-help@sip-communicator.dev.java.net


#9

Eelco Mulder wrote:

No problem, but given all the personal data, I would like to only send
it to a private email address, not a group address...

Which one can I use?

Yes, you can use mine.

Emil

···

Thanks,
Eelco Mulder

2009/9/18 Emil Ivov <emcho@sip-communicator.org
<mailto:emcho@sip-communicator.org>>

    Eelco,

    I only see a CANCEL request in here and this doesn't really tell me that
    much. We would need your complete log file in order to be able to help.

    Emil

    Eelco Mulder wrote:
    > Great, thanks!
    >
    > I have been digging in the logs and I think I found the incoming call
    > entry in the logs. Please let me know what I can read in it or
    what else
    > I can check.... fyi, when I call my SIP number, I hear my pstn phone
    > ringing, but SIP comm does not give me a popup or sound, nor can I
    pick
    > up the phone somehow.
    >
    > Thanks,
    > Eelco Mulder
    >
    > 11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
    > output from the JAIN-SIP stack: semRelease
    > ]]]]gov.nist.javax.sip.stack.SIPServerTransaction@be009483
    > 11:50:33.903 FINER: impl.protocol.sip.SipLogger.logStackTrace().41
    > JAIN-SIP stack trace
    > java.lang.Throwable
    > at
    >
    net.java.sip.communicator.impl.protocol.sip.SipLogger.logStackTrace(SipLogger.java:41)
    > at
    >
    gov.nist.javax.sip.stack.SIPTransaction.semRelease(SIPTransaction.java:1198)
    > at
    >
    gov.nist.javax.sip.stack.SIPTransaction.releaseSem(SIPTransaction.java:1184)
    > at
    >
    gov.nist.javax.sip.stack.SIPServerTransaction.releaseSem(SIPServerTransaction.java:1645)
    > at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:254)
    > at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
    > at java.lang.Thread.run(Unknown Source)
    > 11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
    > output from the JAIN-SIP stack: removePendingTx:
    > z9hg4bk9057a6213f604d23909211763ef079b1
    > 11:50:35.653 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
    > output from the JAIN-SIP stack: UDPMessageChannel:
    > processIncomingDataPacket : peerAddress = 194.120.0.198/5060
    <http://194.120.0.198/5060>
    > <http://194.120.0.198/5060> Length = 537
    > 11:50:35.653 FINE:
    > impl.protocol.sip.AddressResolverImpl.resolveAddress().96
    Returning hop:
    > 123.456.0.999:5060/UDP
    > 11:50:35.653 FINER: impl.protocol.sip.SipLogger.logMessage().188
    > JAIN-SIP received message from "123.456.0.999:5060" to
    "0.0.0.0:5060 <http://0.0.0.0:5060>
    > <http://0.0.0.0:5060>" at 1253267435653
    > CANCEL
    >
    sip:oclee_redlum@10.31.119.53:5060;transport=udp;registering_acc=voipcheap_com
    > SIP/2.0
    > Via: SIP/2.0/UDP
    > 123.456.0.999:5060;branch=z9hL4bK9057a6213f604d23909241763ef079b1
    > From: <sip:0031234128888@voipcheap.com:5060
    <http://sip:0031234128888@voipcheap.com:5060>
    >
    <http://sip:0031234128888@voipcheap.com:5060>>;tag=c11710acc12b10ac4aa6315a20dbd4
    > To: <sip:oclee_redlum@10.31.119.53:5060
    <http://sip:oclee_redlum@10.31.119.53:5060>
    > <http://sip:oclee_redlum@10.31.119.53:5060>>
    > Contact: <sip:0031243128888@194.120.0.198:5060
    <http://sip:0031243128888@194.120.0.198:5060>
    > <http://sip:0031243128888@194.120.0.198:5060>>
    > Call-ID: 2649a604f40b482a8c011393187bc6ac
    > CSeq: 1 CANCEL
    > Server: (Very nice Sip Registrar/Proxy Server)
    > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
    > Content-Length: 0
    >
    > /(please note I have altered IP addresses and telephonenumbers for
    > privacy reasons)/
    >
    >
    > 2009/9/18 Lubomir Marinov <lubomir.marinov@gmail.com
    <mailto:lubomir.marinov@gmail.com>
    > <mailto:lubomir.marinov@gmail.com <mailto:lubomir.marinov@gmail.com>>>
    >
    > Hi Eelco,
    >
    > We just created a FAQ entry at
    >
    http://www.sip-communicator.org/index.php/Documentation/FAQ#logs on
    > the subject of the location of the log files. Please refer to it.
    >
    > Regards,
    > Lubomir
    >
    >
    > On 18.09.2009 11:02, Eelco Mulder wrote:
    >
    > Hi Emil,
    >
    > Thanks for your prompt reply!
    >
    > I'd love to check the logfiles or wireshark dump, but please
    > have some
    > patience with me. I am new to sip comm, so I don't know
    where to
    > find these.
    >
    > I have checked the eventviewer logs, but there are no
    entries in
    > here
    > when I call my SIP number.
    >
    > I have checked the Program File/ Sip Communicator
    directory, but
    > did not
    > find any logging or entry of today.
    >
    > Can you please link me to where and how I can retrieve the
    data?
    >
    > Thanks!
    > Eelco Mulder
    >
    > 2009/9/18 Emil Ivov <emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>
    > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>>>
    >
    >
    > Hey Eelco,
    >
    > Eelco Mulder wrote:
    > > Hi,
    > >
    > > Just to be sure. Has this message arrived the
    correct queue?
    >
    > Yes.
    >
    > > Second, other people can receive inbound sip calls
    without any
    > problems?
    >
    > Yes.
    >
    > > How does this work? (I am using Voipcheap
    mementarily, and
    > inbound calls
    > > with Qutecomm did work flawless).
    >
    > You may want to get a wireshark dump and see how the
    incoming
    > INVITEs
    > look. You can send the dump or a link to it here in
    case you
    > are not
    > able to figure it out by yourself.
    >
    > It would also help to have a look at your log files.
    >
    > Cheers,
    > Emil
    >
    > >
    > > Thanks for your help in advance!
    > >
    > > Regards,
    > > Eelco Mulder
    > >
    > > 2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>>
    > > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>>>>
    >
    > >
    > > Hi,
    > >
    > > I just switched from Qutecomm to SIP
    Communicator. SIP
    > Communicator
    > > looks more stable, even with the current alpha
    > version. Great
    > work!
    > >
    > > But.... (there is always a but)
    > >
    > > Everything works fine, chatting works fine, chat
    > notifcation
    > works
    > > fine, SIP calling works fine.Only, when I receive an
    > incoming
    > call,
    > > nothing happens. No sound, no pop up. I have
    checked the
    > setting is
    > > tool -> notifcations. Both are activated.
    > >
    > > When I look in the call history, only outgoing
    calls are
    > logged...
    > >
    > > What am I missing? Please let me know, I want to
    fully
    > switch
    > to SIP
    > > Communicator.
    > >
    > > Thanks,
    > > Eelco Mulder
    >
    >
    >
    ---------------------------------------------------------------------
    > 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>>
    >
    >

    --
    Emil Ivov, Ph.D. 67000 Strasbourg,
    Project Lead France
    SIP Communicator
    emcho@sip-communicator.org <mailto:emcho@sip-communicator.org>
                  PHONE: +33.1.77.62.43.30
    http://sip-communicator.org FAX: +33.1.77.62.47.31

    ---------------------------------------------------------------------
    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>

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

---------------------------------------------------------------------
To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
For additional commands, e-mail: dev-help@sip-communicator.dev.java.net


#10

Hey Eelco,

It appears that voipcheap.com is sending INVITE requests without a
Max-Forwards header which is mandated by RFC 3261 and it's absence is
hence not allowed by our SIP stack. The requests are therefore ignored.
You may want to report the error there ... or try with another provider.

Hope this helps,
Emil

Emil Ivov wrote:

···

Eelco Mulder wrote:

No problem, but given all the personal data, I would like to only send
it to a private email address, not a group address...

Which one can I use?

Yes, you can use mine.

Emil

Thanks,
Eelco Mulder

2009/9/18 Emil Ivov <emcho@sip-communicator.org
<mailto:emcho@sip-communicator.org>>

    Eelco,

    I only see a CANCEL request in here and this doesn't really tell me that
    much. We would need your complete log file in order to be able to help.

    Emil

    Eelco Mulder wrote:
    > Great, thanks!
    >
    > I have been digging in the logs and I think I found the incoming call
    > entry in the logs. Please let me know what I can read in it or
    what else
    > I can check.... fyi, when I call my SIP number, I hear my pstn phone
    > ringing, but SIP comm does not give me a popup or sound, nor can I
    pick
    > up the phone somehow.
    >
    > Thanks,
    > Eelco Mulder
    >
    > 11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
    > output from the JAIN-SIP stack: semRelease
    > ]]]]gov.nist.javax.sip.stack.SIPServerTransaction@be009483
    > 11:50:33.903 FINER: impl.protocol.sip.SipLogger.logStackTrace().41
    > JAIN-SIP stack trace
    > java.lang.Throwable
    > at
    >
    net.java.sip.communicator.impl.protocol.sip.SipLogger.logStackTrace(SipLogger.java:41)
    > at
    >
    gov.nist.javax.sip.stack.SIPTransaction.semRelease(SIPTransaction.java:1198)
    > at
    >
    gov.nist.javax.sip.stack.SIPTransaction.releaseSem(SIPTransaction.java:1184)
    > at
    >
    gov.nist.javax.sip.stack.SIPServerTransaction.releaseSem(SIPServerTransaction.java:1645)
    > at gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:254)
    > at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
    > at java.lang.Thread.run(Unknown Source)
    > 11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
    > output from the JAIN-SIP stack: removePendingTx:
    > z9hg4bk9057a6213f604d23909211763ef079b1
    > 11:50:35.653 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
    > output from the JAIN-SIP stack: UDPMessageChannel:
    > processIncomingDataPacket : peerAddress = 194.120.0.198/5060
    <http://194.120.0.198/5060>
    > <http://194.120.0.198/5060> Length = 537
    > 11:50:35.653 FINE:
    > impl.protocol.sip.AddressResolverImpl.resolveAddress().96
    Returning hop:
    > 123.456.0.999:5060/UDP
    > 11:50:35.653 FINER: impl.protocol.sip.SipLogger.logMessage().188
    > JAIN-SIP received message from "123.456.0.999:5060" to
    "0.0.0.0:5060 <http://0.0.0.0:5060>
    > <http://0.0.0.0:5060>" at 1253267435653
    > CANCEL
    >
    sip:oclee_redlum@10.31.119.53:5060;transport=udp;registering_acc=voipcheap_com
    > SIP/2.0
    > Via: SIP/2.0/UDP
    > 123.456.0.999:5060;branch=z9hL4bK9057a6213f604d23909241763ef079b1
    > From: <sip:0031234128888@voipcheap.com:5060
    <http://sip:0031234128888@voipcheap.com:5060>
    >
    <http://sip:0031234128888@voipcheap.com:5060>>;tag=c11710acc12b10ac4aa6315a20dbd4
    > To: <sip:oclee_redlum@10.31.119.53:5060
    <http://sip:oclee_redlum@10.31.119.53:5060>
    > <http://sip:oclee_redlum@10.31.119.53:5060>>
    > Contact: <sip:0031243128888@194.120.0.198:5060
    <http://sip:0031243128888@194.120.0.198:5060>
    > <http://sip:0031243128888@194.120.0.198:5060>>
    > Call-ID: 2649a604f40b482a8c011393187bc6ac
    > CSeq: 1 CANCEL
    > Server: (Very nice Sip Registrar/Proxy Server)
    > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
    > Content-Length: 0
    >
    > /(please note I have altered IP addresses and telephonenumbers for
    > privacy reasons)/
    >
    >
    > 2009/9/18 Lubomir Marinov <lubomir.marinov@gmail.com
    <mailto:lubomir.marinov@gmail.com>
    > <mailto:lubomir.marinov@gmail.com <mailto:lubomir.marinov@gmail.com>>>
    >
    > Hi Eelco,
    >
    > We just created a FAQ entry at
    >
    http://www.sip-communicator.org/index.php/Documentation/FAQ#logs on
    > the subject of the location of the log files. Please refer to it.
    >
    > Regards,
    > Lubomir
    >
    >
    > On 18.09.2009 11:02, Eelco Mulder wrote:
    >
    > Hi Emil,
    >
    > Thanks for your prompt reply!
    >
    > I'd love to check the logfiles or wireshark dump, but please
    > have some
    > patience with me. I am new to sip comm, so I don't know
    where to
    > find these.
    >
    > I have checked the eventviewer logs, but there are no
    entries in
    > here
    > when I call my SIP number.
    >
    > I have checked the Program File/ Sip Communicator
    directory, but
    > did not
    > find any logging or entry of today.
    >
    > Can you please link me to where and how I can retrieve the
    data?
    >
    > Thanks!
    > Eelco Mulder
    >
    > 2009/9/18 Emil Ivov <emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>
    > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>>>
    >
    >
    > Hey Eelco,
    >
    > Eelco Mulder wrote:
    > > Hi,
    > >
    > > Just to be sure. Has this message arrived the
    correct queue?
    >
    > Yes.
    >
    > > Second, other people can receive inbound sip calls
    without any
    > problems?
    >
    > Yes.
    >
    > > How does this work? (I am using Voipcheap
    mementarily, and
    > inbound calls
    > > with Qutecomm did work flawless).
    >
    > You may want to get a wireshark dump and see how the
    incoming
    > INVITEs
    > look. You can send the dump or a link to it here in
    case you
    > are not
    > able to figure it out by yourself.
    >
    > It would also help to have a look at your log files.
    >
    > Cheers,
    > Emil
    >
    > >
    > > Thanks for your help in advance!
    > >
    > > Regards,
    > > Eelco Mulder
    > >
    > > 2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>>
    > > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>>>>
    >
    > >
    > > Hi,
    > >
    > > I just switched from Qutecomm to SIP
    Communicator. SIP
    > Communicator
    > > looks more stable, even with the current alpha
    > version. Great
    > work!
    > >
    > > But.... (there is always a but)
    > >
    > > Everything works fine, chatting works fine, chat
    > notifcation
    > works
    > > fine, SIP calling works fine.Only, when I receive an
    > incoming
    > call,
    > > nothing happens. No sound, no pop up. I have
    checked the
    > setting is
    > > tool -> notifcations. Both are activated.
    > >
    > > When I look in the call history, only outgoing
    calls are
    > logged...
    > >
    > > What am I missing? Please let me know, I want to
    fully
    > switch
    > to SIP
    > > Communicator.
    > >
    > > Thanks,
    > > Eelco Mulder
    >
    >
    >
    ---------------------------------------------------------------------
    > 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>>
    >
    >

    --
    Emil Ivov, Ph.D. 67000 Strasbourg,
    Project Lead France
    SIP Communicator
    emcho@sip-communicator.org <mailto:emcho@sip-communicator.org>
                  PHONE: +33.1.77.62.43.30
    http://sip-communicator.org FAX: +33.1.77.62.47.31

    ---------------------------------------------------------------------
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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>

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

---------------------------------------------------------------------
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#11

Hi Emil.

Thanks for checking and finding the reason!

Voipcheap is one of the voipbuster companies and one of most used voip
providers, at least in Europe.They provide their own sip program, so I can
almost be 100% sure they won´t change the protocol, but I wil try.

It will be a main set back for sip comm when these voipbuster provider(s)
cannot be used. People will use Qutecomm (Wengophone) or other sip progs
instead I guess.

There is no chance to build this as a plugin?

Regards,
Eelco Mulder

···

2009/9/18 Emil Ivov <emcho@sip-communicator.org>

Hey Eelco,

It appears that voipcheap.com is sending INVITE requests without a
Max-Forwards header which is mandated by RFC 3261 and it's absence is
hence not allowed by our SIP stack. The requests are therefore ignored.
You may want to report the error there ... or try with another provider.

Hope this helps,
Emil

Emil Ivov wrote:
> Eelco Mulder wrote:
>> No problem, but given all the personal data, I would like to only send
>> it to a private email address, not a group address...
>>
>> Which one can I use?
>
> Yes, you can use mine.
>
> Emil
>> Thanks,
>> Eelco Mulder
>>
>> 2009/9/18 Emil Ivov <emcho@sip-communicator.org
>> <mailto:emcho@sip-communicator.org>>
>>
>> Eelco,
>>
>> I only see a CANCEL request in here and this doesn't really tell me
that
>> much. We would need your complete log file in order to be able to
help.
>>
>> Emil
>>
>> Eelco Mulder wrote:
>> > Great, thanks!
>> >
>> > I have been digging in the logs and I think I found the incoming
call
>> > entry in the logs. Please let me know what I can read in it or
>> what else
>> > I can check.... fyi, when I call my SIP number, I hear my pstn
phone
>> > ringing, but SIP comm does not give me a popup or sound, nor can I
>> pick
>> > up the phone somehow.
>> >
>> > Thanks,
>> > Eelco Mulder
>> >
>> > 11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
>> > output from the JAIN-SIP stack: semRelease
>> > ]]]]gov.nist.javax.sip.stack.SIPServerTransaction@be009483
>> > 11:50:33.903 FINER: impl.protocol.sip.SipLogger.logStackTrace().41
>> > JAIN-SIP stack trace
>> > java.lang.Throwable
>> > at
>> >
>>
net.java.sip.communicator.impl.protocol.sip.SipLogger.logStackTrace(SipLogger.java:41)
>> > at
>> >
>>
gov.nist.javax.sip.stack.SIPTransaction.semRelease(SIPTransaction.java:1198)
>> > at
>> >
>>
gov.nist.javax.sip.stack.SIPTransaction.releaseSem(SIPTransaction.java:1184)
>> > at
>> >
>>
gov.nist.javax.sip.stack.SIPServerTransaction.releaseSem(SIPServerTransaction.java:1645)
>> > at
gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:254)
>> > at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
>> > at java.lang.Thread.run(Unknown Source)
>> > 11:50:33.903 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
>> > output from the JAIN-SIP stack: removePendingTx:
>> > z9hg4bk9057a6213f604d23909211763ef079b1
>> > 11:50:35.653 FINE: impl.protocol.sip.SipLogger.logDebug().78 Debug
>> > output from the JAIN-SIP stack: UDPMessageChannel:
>> > processIncomingDataPacket : peerAddress = 194.120.0.198/5060
>> <http://194.120.0.198/5060>
>> > <http://194.120.0.198/5060> Length = 537
>> > 11:50:35.653 FINE:
>> > impl.protocol.sip.AddressResolverImpl.resolveAddress().96
>> Returning hop:
>> > 123.456.0.999:5060/UDP
>> > 11:50:35.653 FINER: impl.protocol.sip.SipLogger.logMessage().188
>> > JAIN-SIP received message from "123.456.0.999:5060" to
>> "0.0.0.0:5060 <http://0.0.0.0:5060>
>> > <http://0.0.0.0:5060>" at 1253267435653
>> > CANCEL
>> >
>> sip:oclee_redlum@10.31.119.53:5060
;transport=udp;registering_acc=voipcheap_com
>> > SIP/2.0
>> > Via: SIP/2.0/UDP
>> > 123.456.0.999:5060;branch=z9hL4bK9057a6213f604d23909241763ef079b1
>> > From: <sip:0031234128888@voipcheap.com:5060
>> <http://sip:0031234128888@voipcheap.com:5060>
>> >
>> <http://sip:0031234128888@voipcheap.com:5060
>>;tag=c11710acc12b10ac4aa6315a20dbd4
>> > To: <sip:oclee_redlum@10.31.119.53:5060
>> <http://sip:oclee_redlum@10.31.119.53:5060>
>> > <http://sip:oclee_redlum@10.31.119.53:5060>>
>> > Contact: <sip:0031243128888@194.120.0.198:5060
>> <http://sip:0031243128888@194.120.0.198:5060>
>> > <http://sip:0031243128888@194.120.0.198:5060>>
>> > Call-ID: 2649a604f40b482a8c011393187bc6ac
>> > CSeq: 1 CANCEL
>> > Server: (Very nice Sip Registrar/Proxy Server)
>> > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
>> > Content-Length: 0
>> >
>> > /(please note I have altered IP addresses and telephonenumbers for
>> > privacy reasons)/
>> >
>> >
>> > 2009/9/18 Lubomir Marinov <lubomir.marinov@gmail.com
>> <mailto:lubomir.marinov@gmail.com>
>> > <mailto:lubomir.marinov@gmail.com <mailto:
lubomir.marinov@gmail.com>>>
>> >
>> > Hi Eelco,
>> >
>> > We just created a FAQ entry at
>> >
>> http://www.sip-communicator.org/index.php/Documentation/FAQ#logs on
>> > the subject of the location of the log files. Please refer to
it.
>> >
>> > Regards,
>> > Lubomir
>> >
>> >
>> > On 18.09.2009 11:02, Eelco Mulder wrote:
>> >
>> > Hi Emil,
>> >
>> > Thanks for your prompt reply!
>> >
>> > I'd love to check the logfiles or wireshark dump, but
please
>> > have some
>> > patience with me. I am new to sip comm, so I don't know
>> where to
>> > find these.
>> >
>> > I have checked the eventviewer logs, but there are no
>> entries in
>> > here
>> > when I call my SIP number.
>> >
>> > I have checked the Program File/ Sip Communicator
>> directory, but
>> > did not
>> > find any logging or entry of today.
>> >
>> > Can you please link me to where and how I can retrieve the
>> data?
>> >
>> > Thanks!
>> > Eelco Mulder
>> >
>> > 2009/9/18 Emil Ivov <emcho@sip-communicator.org
>> <mailto:emcho@sip-communicator.org>
>> > <mailto:emcho@sip-communicator.org
>> <mailto:emcho@sip-communicator.org>>
>> > <mailto:emcho@sip-communicator.org
>> <mailto:emcho@sip-communicator.org>
>> > <mailto:emcho@sip-communicator.org
>> <mailto:emcho@sip-communicator.org>>>>
>> >
>> >
>> > Hey Eelco,
>> >
>> > Eelco Mulder wrote:
>> > > Hi,
>> > >
>> > > Just to be sure. Has this message arrived the
>> correct queue?
>> >
>> > Yes.
>> >
>> > > Second, other people can receive inbound sip calls
>> without any
>> > problems?
>> >
>> > Yes.
>> >
>> > > How does this work? (I am using Voipcheap
>> mementarily, and
>> > inbound calls
>> > > with Qutecomm did work flawless).
>> >
>> > You may want to get a wireshark dump and see how the
>> incoming
>> > INVITEs
>> > look. You can send the dump or a link to it here in
>> case you
>> > are not
>> > able to figure it out by yourself.
>> >
>> > It would also help to have a look at your log files.
>> >
>> > Cheers,
>> > Emil
>> >
>> > >
>> > > Thanks for your help in advance!
>> > >
>> > > Regards,
>> > > Eelco Mulder
>> > >
>> > > 2009/9/17 Eelco Mulder <eelco_mulder@hotmail.com
>> <mailto:eelco_mulder@hotmail.com>
>> > <mailto:eelco_mulder@hotmail.com
>> <mailto:eelco_mulder@hotmail.com>>
>> > <mailto:eelco_mulder@hotmail.com
>> <mailto:eelco_mulder@hotmail.com>
>> > <mailto:eelco_mulder@hotmail.com
>> <mailto:eelco_mulder@hotmail.com>>>
>> > > <mailto:eelco_mulder@hotmail.com
>> <mailto:eelco_mulder@hotmail.com>
>> > <mailto:eelco_mulder@hotmail.com
>> <mailto:eelco_mulder@hotmail.com>>
>> > <mailto:eelco_mulder@hotmail.com
>> <mailto:eelco_mulder@hotmail.com>
>> > <mailto:eelco_mulder@hotmail.com
>> <mailto:eelco_mulder@hotmail.com>>>>>
>> >
>> > >
>> > > Hi,
>> > >
>> > > I just switched from Qutecomm to SIP
>> Communicator. SIP
>> > Communicator
>> > > looks more stable, even with the current alpha
>> > version. Great
>> > work!
>> > >
>> > > But.... (there is always a but)
>> > >
>> > > Everything works fine, chatting works fine, chat
>> > notifcation
>> > works
>> > > fine, SIP calling works fine.Only, when I
receive an
>> > incoming
>> > call,
>> > > nothing happens. No sound, no pop up. I have
>> checked the
>> > setting is
>> > > tool -> notifcations. Both are activated.
>> > >
>> > > When I look in the call history, only outgoing
>> calls are
>> > logged...
>> > >
>> > > What am I missing? Please let me know, I want to
>> fully
>> > switch
>> > to SIP
>> > > Communicator.
>> > >
>> > > Thanks,
>> > > Eelco Mulder
>> >
>> >
>> >
>>
---------------------------------------------------------------------
>> > 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>>
>> >
>> >
>>
>> --
>> Emil Ivov, Ph.D. 67000 Strasbourg,
>> Project Lead France
>> SIP Communicator
>> emcho@sip-communicator.org <mailto:emcho@sip-communicator.org>
>> PHONE: +33.1.77.62.43.30
>> http://sip-communicator.org FAX:
+33.1.77.62.47.31
>>
>>
---------------------------------------------------------------------
>> 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>
>>
>>
>

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

---------------------------------------------------------------------
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#12

Hello Eelco,

Eelco Mulder wrote:

Hi Emil.

Thanks for checking and finding the reason!

Voipcheap is one of the voipbuster companies and one of most used voip
providers, at least in Europe.They provide their own sip program, so I
can almost be 100% sure they won´t change the protocol, but I wil try.

They don't need to change the protocol but respect it (the Max-Forwards
header is mandated for all requests by RFC 3261). I doubt that
respecting the RFC is going to break their existing client. Anyways, if
they don't, SIP Communicator is not going to be the only agent that they
are going to have problems with, so unless they are explicitly trying to
restrict access to their own client (in which case there's nothing we
could do) then they should take care of this.

It will be a main set back for sip comm when these voipbuster
provider(s) cannot be used.

Frankly, this is the first provider where I see this and SIP
Communicator works with most that I've tried.

People will use Qutecomm (Wengophone) or
other sip progs instead I guess.

There is no chance to build this as a plugin?

Well, plugins are normally used to add support for new features or
protocols. In this case we have non-respect of an existing standardized
protocol. We couldn't possibly adapt to the bugs and idiosyncrasies of
all providers around the world. The very point of standards and RFCs is
to avoid this.

The best thing to do here is to simply contact them and see whether
they'd be willing to fix the problem. Let us know how it goes.

Cheers,
Emil

···

Regards,
Eelco Mulder

2009/9/18 Emil Ivov <emcho@sip-communicator.org
<mailto:emcho@sip-communicator.org>>

    Hey Eelco,

    It appears that voipcheap.com <http://voipcheap.com> is sending
    INVITE requests without a
    Max-Forwards header which is mandated by RFC 3261 and it's absence is
    hence not allowed by our SIP stack. The requests are therefore ignored.
    You may want to report the error there ... or try with another provider.

    Hope this helps,
    Emil

    Emil Ivov wrote:
    > Eelco Mulder wrote:
    >> No problem, but given all the personal data, I would like to only
    send
    >> it to a private email address, not a group address...
    >>
    >> Which one can I use?
    >
    > Yes, you can use mine.
    >
    > Emil
    >> Thanks,
    >> Eelco Mulder
    >>
    >> 2009/9/18 Emil Ivov <emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    >> <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>>
    >>
    >> Eelco,
    >>
    >> I only see a CANCEL request in here and this doesn't really
    tell me that
    >> much. We would need your complete log file in order to be
    able to help.
    >>
    >> Emil
    >>
    >> Eelco Mulder wrote:
    >> > Great, thanks!
    >> >
    >> > I have been digging in the logs and I think I found the
    incoming call
    >> > entry in the logs. Please let me know what I can read in it or
    >> what else
    >> > I can check.... fyi, when I call my SIP number, I hear my
    pstn phone
    >> > ringing, but SIP comm does not give me a popup or sound,
    nor can I
    >> pick
    >> > up the phone somehow.
    >> >
    >> > Thanks,
    >> > Eelco Mulder
    >> >
    >> > 11:50:33.903 FINE:
    impl.protocol.sip.SipLogger.logDebug().78 Debug
    >> > output from the JAIN-SIP stack: semRelease
    >> > ]]]]gov.nist.javax.sip.stack.SIPServerTransaction@be009483
    >> > 11:50:33.903 FINER:
    impl.protocol.sip.SipLogger.logStackTrace().41
    >> > JAIN-SIP stack trace
    >> > java.lang.Throwable
    >> > at
    >> >
    >>
    net.java.sip.communicator.impl.protocol.sip.SipLogger.logStackTrace(SipLogger.java:41)
    >> > at
    >> >
    >>
    gov.nist.javax.sip.stack.SIPTransaction.semRelease(SIPTransaction.java:1198)
    >> > at
    >> >
    >>
    gov.nist.javax.sip.stack.SIPTransaction.releaseSem(SIPTransaction.java:1184)
    >> > at
    >> >
    >>
    gov.nist.javax.sip.stack.SIPServerTransaction.releaseSem(SIPServerTransaction.java:1645)
    >> > at
    gov.nist.javax.sip.EventScanner.deliverEvent(EventScanner.java:254)
    >> > at gov.nist.javax.sip.EventScanner.run(EventScanner.java:492)
    >> > at java.lang.Thread.run(Unknown Source)
    >> > 11:50:33.903 FINE:
    impl.protocol.sip.SipLogger.logDebug().78 Debug
    >> > output from the JAIN-SIP stack: removePendingTx:
    >> > z9hg4bk9057a6213f604d23909211763ef079b1
    >> > 11:50:35.653 FINE:
    impl.protocol.sip.SipLogger.logDebug().78 Debug
    >> > output from the JAIN-SIP stack: UDPMessageChannel:
    >> > processIncomingDataPacket : peerAddress =
    194.120.0.198/5060 <http://194.120.0.198/5060>
    >> <http://194.120.0.198/5060>
    >> > <http://194.120.0.198/5060> Length = 537
    >> > 11:50:35.653 FINE:
    >> > impl.protocol.sip.AddressResolverImpl.resolveAddress().96
    >> Returning hop:
    >> > 123.456.0.999:5060/UDP
    >> > 11:50:35.653 FINER:
    impl.protocol.sip.SipLogger.logMessage().188
    >> > JAIN-SIP received message from "123.456.0.999:5060" to
    >> "0.0.0.0:5060 <http://0.0.0.0:5060> <http://0.0.0.0:5060>
    >> > <http://0.0.0.0:5060>" at 1253267435653
    >> > CANCEL
    >> >
    >>
    sip:oclee_redlum@10.31.119.53:5060;transport=udp;registering_acc=voipcheap_com
    >> > SIP/2.0
    >> > Via: SIP/2.0/UDP
    >> >
    123.456.0.999:5060;branch=z9hL4bK9057a6213f604d23909241763ef079b1
    >> > From: <sip:0031234128888@voipcheap.com:5060
    <http://sip:0031234128888@voipcheap.com:5060>
    >> <http://sip:0031234128888@voipcheap.com:5060>
    >> >
    >>
    <http://sip:0031234128888@voipcheap.com:5060>>;tag=c11710acc12b10ac4aa6315a20dbd4
    >> > To: <sip:oclee_redlum@10.31.119.53:5060
    <http://sip:oclee_redlum@10.31.119.53:5060>
    >> <http://sip:oclee_redlum@10.31.119.53:5060>
    >> > <http://sip:oclee_redlum@10.31.119.53:5060>>
    >> > Contact: <sip:0031243128888@194.120.0.198:5060
    <http://sip:0031243128888@194.120.0.198:5060>
    >> <http://sip:0031243128888@194.120.0.198:5060>
    >> > <http://sip:0031243128888@194.120.0.198:5060>>
    >> > Call-ID: 2649a604f40b482a8c011393187bc6ac
    >> > CSeq: 1 CANCEL
    >> > Server: (Very nice Sip Registrar/Proxy Server)
    >> > Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
    >> > Content-Length: 0
    >> >
    >> > /(please note I have altered IP addresses and
    telephonenumbers for
    >> > privacy reasons)/
    >> >
    >> >
    >> > 2009/9/18 Lubomir Marinov <lubomir.marinov@gmail.com
    <mailto:lubomir.marinov@gmail.com>
    >> <mailto:lubomir.marinov@gmail.com
    <mailto:lubomir.marinov@gmail.com>>
    >> > <mailto:lubomir.marinov@gmail.com
    <mailto:lubomir.marinov@gmail.com> <mailto:lubomir.marinov@gmail.com
    <mailto:lubomir.marinov@gmail.com>>>>
    >> >
    >> > Hi Eelco,
    >> >
    >> > We just created a FAQ entry at
    >> >
    >>
    http://www.sip-communicator.org/index.php/Documentation/FAQ#logs on
    >> > the subject of the location of the log files. Please
    refer to it.
    >> >
    >> > Regards,
    >> > Lubomir
    >> >
    >> >
    >> > On 18.09.2009 11:02, Eelco Mulder wrote:
    >> >
    >> > Hi Emil,
    >> >
    >> > Thanks for your prompt reply!
    >> >
    >> > I'd love to check the logfiles or wireshark dump,
    but please
    >> > have some
    >> > patience with me. I am new to sip comm, so I don't know
    >> where to
    >> > find these.
    >> >
    >> > I have checked the eventviewer logs, but there are no
    >> entries in
    >> > here
    >> > when I call my SIP number.
    >> >
    >> > I have checked the Program File/ Sip Communicator
    >> directory, but
    >> > did not
    >> > find any logging or entry of today.
    >> >
    >> > Can you please link me to where and how I can
    retrieve the
    >> data?
    >> >
    >> > Thanks!
    >> > Eelco Mulder
    >> >
    >> > 2009/9/18 Emil Ivov <emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    >> <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>
    >> > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    >> <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>>
    >> > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    >> <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>
    >> > <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    >> <mailto:emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>>>>>
    >> >
    >> >
    >> > Hey Eelco,
    >> >
    >> > Eelco Mulder wrote:
    >> > > Hi,
    >> > >
    >> > > Just to be sure. Has this message arrived the
    >> correct queue?
    >> >
    >> > Yes.
    >> >
    >> > > Second, other people can receive inbound sip
    calls
    >> without any
    >> > problems?
    >> >
    >> > Yes.
    >> >
    >> > > How does this work? (I am using Voipcheap
    >> mementarily, and
    >> > inbound calls
    >> > > with Qutecomm did work flawless).
    >> >
    >> > You may want to get a wireshark dump and see how the
    >> incoming
    >> > INVITEs
    >> > look. You can send the dump or a link to it here in
    >> case you
    >> > are not
    >> > able to figure it out by yourself.
    >> >
    >> > It would also help to have a look at your log files.
    >> >
    >> > Cheers,
    >> > Emil
    >> >
    >> > >
    >> > > Thanks for your help in advance!
    >> > >
    >> > > Regards,
    >> > > Eelco Mulder
    >> > >
    >> > > 2009/9/17 Eelco Mulder
    <eelco_mulder@hotmail.com <mailto:eelco_mulder@hotmail.com>
    >> <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>
    >> > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    >> <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>>
    >> > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    >> <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>
    >> > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    >> <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>>>
    >> > > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    >> <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>
    >> > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    >> <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>>
    >> > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    >> <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>
    >> > <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>
    >> <mailto:eelco_mulder@hotmail.com
    <mailto:eelco_mulder@hotmail.com>>>>>>
    >> >
    >> > >
    >> > > Hi,
    >> > >
    >> > > I just switched from Qutecomm to SIP
    >> Communicator. SIP
    >> > Communicator
    >> > > looks more stable, even with the current
    alpha
    >> > version. Great
    >> > work!
    >> > >
    >> > > But.... (there is always a but)
    >> > >
    >> > > Everything works fine, chatting works
    fine, chat
    >> > notifcation
    >> > works
    >> > > fine, SIP calling works fine.Only, when I
    receive an
    >> > incoming
    >> > call,
    >> > > nothing happens. No sound, no pop up. I have
    >> checked the
    >> > setting is
    >> > > tool -> notifcations. Both are activated.
    >> > >
    >> > > When I look in the call history, only
    outgoing
    >> calls are
    >> > logged...
    >> > >
    >> > > What am I missing? Please let me know, I
    want to
    >> fully
    >> > switch
    >> > to SIP
    >> > > Communicator.
    >> > >
    >> > > Thanks,
    >> > > Eelco Mulder
    >> >
    >> >
    >> >
    >>
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    <mailto:dev-unsubscribe@sip-communicator.dev.java.net>>>
    >> > 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
    <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>>>
    >> >
    >> >
    >>
    >> --
    >> Emil Ivov, Ph.D. 67000 Strasbourg,
    >> Project Lead France
    >> SIP Communicator
    >> emcho@sip-communicator.org
    <mailto:emcho@sip-communicator.org>
    <mailto:emcho@sip-communicator.org <mailto:emcho@sip-communicator.org>>
    >> PHONE: +33.1.77.62.43.30
    >> http://sip-communicator.org FAX:
    +33.1.77.62.47.31
    >>
    >>
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    <mailto:dev-help@sip-communicator.dev.java.net>>
    >>
    >>
    >

    --
    Emil Ivov, Ph.D. 67000 Strasbourg,
    Project Lead France
    SIP Communicator
    emcho@sip-communicator.org <mailto:emcho@sip-communicator.org>
                  PHONE: +33.1.77.62.43.30
    http://sip-communicator.org FAX: +33.1.77.62.47.31

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--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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