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        <rdf:li rdf:resource="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12891"/>
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        <rdf:li rdf:resource="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12885"/>
        <rdf:li rdf:resource="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12879"/>
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    <link>http://gmane.org</link>
  </image>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12937">
    <title>Fax through sipxbridge issue -- re-INIVTE changesreceive port.</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12937</link>
    <description>Hello,

I am testing sending of FAX via sipxbridge. I find that the Audiocodes
INVITEs using G711 offer and then Re-INVITEs using a t38 offer but now
with a different receive port. However, the session ID for the SDP is
the same. Does this mean there are actually separate receive ports
associated with the same session?

Attached is the merged.xml  which would clarify the situation.

Note on Frame 1, Audiocodes wants to receive on Port 6000
Then on Frame 28 Audiocodes re-INVITES within the same dialog but now
it wants to receive on port 6002.

What happens to the data that was destined to port 6000 at this point?

I ask because in the context of sipxbridge, I want to know whether or
not to create a second bridge for this port.

Second, do I need RTCP support for fax?

Thank you!


Ranga

</description>
    <dc:creator>M. Ranganathan</dc:creator>
    <dc:date>2008-10-07T16:31:43</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12934">
    <title>DTMF transport incompatibility</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12934</link>
    <description>One of our frequent problems are reports that Voicemail or AA does not
respond correctly to DTMF; these nearly always turn out to be that the
sender is not sending the DTMF using telephone-event encoding (RFC 2833
or its successor) in the RTP.

Since this failure could be detected at the INVITE when we see that the
SDP does not support the only DTMF transport we support, shouldn't we
just reject these calls with a some 4xx media unsupported response code
and a nice informative message explaining the problem?



</description>
    <dc:creator>Scott Lawrence</dc:creator>
    <dc:date>2008-10-07T15:00:32</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12931">
    <title>not able to get High Availability implementation</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12931</link>
    <description/>
    <dc:creator>Vikas Sharma</dc:creator>
    <dc:date>2008-10-07T11:07:52</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12930">
    <title>ITSP test</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12930</link>
    <description>Hi - I´m looking to test sip trunking with an itsp that i use in Europe.

I´d be looking to set all components on the same server using ISO 
what would be the best build to use?  3.11.4?


</description>
    <dc:creator>xavier houghton</dc:creator>
    <dc:date>2008-10-07T10:56:52</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12923">
    <title>new alarm e-mail notification table is non-standard</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12923</link>
    <description>This table seems to be misleading. You press "Enable e-mail notifications"
and it if the checkbox is not checked it actually disables notification.

I think it would be better if e-mail notification column had
"Disable/Enable" toggle and was saved with the same "Apply" button as
everything else.

If we want to keep Checkbox column we need two buttons below the table:
"Enable e-mail notification" and "Disable e-mail nofication". (but I like
it a lot less then a drop-down toggle in each row).

In sipXconfig buttons below the table have to affect all "selected" rows in
a way consistent with a button label. We cannot have a button affect the
row differently based on the selection status.

D.

</description>
    <dc:creator>Damian Krzeminski</dc:creator>
    <dc:date>2008-10-06T19:36:34</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12921">
    <title>process mgmt: dependent processes etc</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12921</link>
    <description/>
    <dc:creator>Carolyn Beeton</dc:creator>
    <dc:date>2008-10-06T18:40:55</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12912">
    <title>DNS configuration in High Availability</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12912</link>
    <description/>
    <dc:creator>Vikas Sharma</dc:creator>
    <dc:date>2008-10-06T10:33:31</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12910">
    <title>SipxBridge Hairpinned in bound calls originating at pbx-- why LOOP DETECTED?</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12910</link>
    <description>Hello,

I am testing hairpinned inbound calls from the PBX via a siptrunk back
to the pbx. Following is the scenario:

- Extension 202 dials the ITSP DID number that is assigned to my pbx.
- Extension 201 gets the call.
- Extension 202 puts the call on hold. MOH plays at extension 201
- Extension 202 removes the hold.  So far all is fine and the behavior
is as expected.  Now I have trouble...
- Extension 202 puts call on hold again. I get LOOP DETECTED from extension 201

Attached is the mreged.xml. for this scenario. Note the LOOP DETECTED
on Frame 141.

I am unable to tell why this is happening based on the signaling
especially because it appears to be exactly the same signaling as the
first hold/unhold attempt.

If you have time to look at the attached trace, perhaps I missed something.

My version is 3.11.6-013602

Thanks in advance.

Regards,

Ranga



</description>
    <dc:creator>M. Ranganathan</dc:creator>
    <dc:date>2008-10-06T04:07:54</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12909">
    <title>test</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12909</link>
    <description>checking list no messages since 10am

</description>
    <dc:creator>Kathleen Eccles</dc:creator>
    <dc:date>2008-10-06T03:35:46</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12908">
    <title>How To solve some fails jobs during sipxpbx config testrelatedwith sipxregistrar, sipxproxy and sipxpage?</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12908</link>
    <description>Hi,

i have installed the latest unstable version of sipxecs on our centos 5
system. I'm trying to configure it to connect to a voipuser.org sip
account. But when i did configtest i get these fails:

/etc/init.d/sipxpbx configtest
Checking TLS/SSL configuration:[  OK  ]
Checking Per-process file descriptor limits:[  OK  ]
Checking rpm configuration file updates:[  OK  ]
Checking SELinux is not enforcing:[  OK  ]
Checking Apache configuration:[  OK  ]
Checking hostname is fully qualified:[  OK  ]
Checking localhost address configured:[  OK  ]
Checking localhost name is not shared:[  OK  ]
Checking configuration service hostname and address:[  OK  ]
Checking /tmp directory has correct permissions:[  OK  ]
Checking sipxsupervisor:[  OK  ]
Checking freeswitch:[  OK  ]
Checking sipregistrar:[FAILED]
Checking sipstatus:[  OK  ]
Checking sipxacd:[  OK  ]
Checking sipxbridge:[  OK  ]
Checking sipxcallresolver-agent: (Disabled) [  OK  ]
Checking sipxcallresolver:[  OK  ]
Checking sipxconfig-agent:[  OK  ]
Checking sipxconfig:[  OK  ]
Checking sipxpage:[FAILED]
Checking sipxpark:[  OK  ]
Checking sipxpresence:[  OK  ]
Checking sipXproxy:[FAILED]
Checking sipxrelay:[  OK  ]
Checking sipxrls:[  OK  ]
Checking sipXvxml:[  OK  ]
sipXpbx: 
sipXpbx: sipXpbx configuration problems found:
sipXpbx: 
sipXpbx: Check sipregistrar
sipXpbx:   Configuration file not found: '/etc/sipxpbx/mappingrules.xml'
sipXpbx:   Configuration file not found:
'/etc/sipxpbx/fallbackrules.xml'
sipXpbx: 
sipXpbx: Check sipxpage
sipXpbx:   Configuration file not found:
'/etc/sipxpbx/sipxpage.properties'
sipXpbx: 
sipXpbx: Check sipXproxy
sipXpbx:   Configuration file not found:
'/etc/sipxpbx/forwardingrules.xml'
sipXpbx:   Configuration file not found: '/etc/sipxpbx/authrules.xml'
sipXpbx:   
sipXpbx:   Error at file /etc/sipxpbx/nattraversalrules.xml, line 21,
char 10
sipXpbx:     Message: Element 'publicport' is not valid for content
model:
'((state,behindnat,useSTUN,stun-server-address,rediscovery-time,publicaddress,publicport,proxyhostport,relayaggressiveness,concurrentrelays,mediarelayexternaladdress,mediarelaynativeaddress,mediarelayxml-rpc-port,port-range,log-level),log-directory)'
sipXpbx: 


How to solve them?

also i get these fails in jobstatus page in the interface:

Data replication: extension Failed
File replication: validusers.xml Failed

About configuration i followed:
http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration

Do i need to do something else to get user with extension 240 to
register to sipxecs and make calls thru sipxecs?

But i want first to solve the problems with failed jobs, then i can
investigate more for registering to the voipuser.org and registering
extensions to sipxecs.

I will be happy if someone help me to get sipxecs working.

Thanks in advanced!

Regards, Ali Nebi!

</description>
    <dc:creator>Ali Nebi</dc:creator>
    <dc:date>2008-10-05T08:25:29</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12907">
    <title>Problem in creating SSL certificates for distributedserver in HA</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12907</link>
    <description/>
    <dc:creator>Vikas Sharma</dc:creator>
    <dc:date>2008-10-04T11:13:10</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12904">
    <title>sip&lt;process&gt;.sh scripts don't all --stop successfully</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12904</link>
    <description/>
    <dc:creator>Carolyn Beeton</dc:creator>
    <dc:date>2008-10-03T18:35:33</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12899">
    <title>SipXConfig: Inserting a Link into User Phones page onSCS 500 Web portal</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12899</link>
    <description/>
    <dc:creator>Jiale Huo</dc:creator>
    <dc:date>2008-10-03T15:00:39</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12897">
    <title>Query Regarding ACD Message Handling?</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12897</link>
    <description>Hi All,

         I have a query releated to  ACD server working of Sipxces
regrading this I have done couple of tasks

         First of all, I added two phones in a queue to act as agents
300 and 301 and then I made a call from a outer phone 400.

         After that, I checked logs of ACD server in "acd log file",
also checked logs of Proxy Server in "proxy log file".

         I found that acd log file generates the log in the form of
sipxacd:"INVITE", so I want to know that how it distinguish this
particular SIP message is for
        ACD server. Is sipxces distinguish anywhere between general
Sipxces message and ACD server message.

        Please, reply me it is urgent.

Thanks In Advance,
Anand.
</description>
    <dc:creator>Anand Yogas</dc:creator>
    <dc:date>2008-10-03T14:10:04</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12893">
    <title>Displaying Active calls on web page</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12893</link>
    <description/>
    <dc:creator>Vikas Sharma</dc:creator>
    <dc:date>2008-10-03T07:22:14</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12891">
    <title>ITSP testing</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12891</link>
    <description/>
    <dc:creator>Yakout Esmat</dc:creator>
    <dc:date>2008-10-02T22:02:33</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12887">
    <title>Problems with c++ projects on PPC64 platform with FC9</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12887</link>
    <description>In order to test freeswitch based components on the PPC64 platform (for 
XECS-1441), I updated my system to FC9.  There appears to be a problem with 
building C++ projects on this platform.  Specifically, the checking done by 
configure fails when it attempts to check for the default output file name 
generated by the C++ compiler.  Here is the output:
    checking for C++ compiler default output file name...
    configure: error: C++ compiler cannot create executables
    See `config.log' for more details.
The file config.log contains no useful information with regard to the 
problem.

Further investigation revealed that the C++ compiler flag -m64 causes this 
problem.  When this flag is specified, the compiler (GCC 4.3.0-8) generates 
code that is considered by ld incompatible with several libraries.  Here is 
the output:
    /usr/bin/ld: skipping incompatible 
/usr/lib/gcc/ppc64-redhat-linux/4.3.0/../../../libc.so when searching 
for -lc
    /usr/bin/ld: skipping incompatible 
/usr/lib/gcc/ppc64-redhat-linux/4.3.0/../../../libc.a when searching for -lc
    /usr/bin/ld: skipping incompatible /usr/lib/libc.so when searching 
for -lc
    /usr/bin/ld: skipping incompatible /usr/lib/libc.a when searching 
for -lc
    /usr/bin/ld: cannot find -lc
    collect2: ld returned 1 exit status

It appears that this could be a problem with either gcc, ld or libraries. 
Removing the -m64 flag could be the simple way to proceed - I don't know 
though if that would be correct and also I am not sure how to automatically 
generate the C++ flags for our projects without this flag specified.  Trying 
to determine where the compatibility problem really is does not appear to be 
terribly simple either.  I would appreciate any suggestion how to resolve 
this issue.

Thanks,
Misha

P.S. To reproduce the issue, all what is required is to compile a simple C++ 
executable with the -m64 flag specified. 



</description>
    <dc:creator>Misha Vodsedalek</dc:creator>
    <dc:date>2008-10-02T21:11:05</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12885">
    <title>Also having errors about httpd-sipxchange-common.confand httpd-sipxchange-common-ssl.conf</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12885</link>
    <description>Martin sent an email about this the other day but there was no reply. 
Now I am having the same problem. I'm trying to get sipX up and running 
on a Fedora 8 virtual machine. It builds and installs without a hitch, 
but when I try to start it, I get the same problem Martin described:

Starting httpd: httpd: Syntax error on line 1049 of 
/usr/local/sipx/etc/sipxpbx/httpd.conf: Could not open configuration 
file /usr/local/sipx/etc/sipxpbx/httpd-sipxchange-common-ssl.conf: No 
such file or directory

This is on Fedora 8, on the latest (svn 13583).

Any ideas what's up here? There are corresponding .in files in 
/usr/local/sipx/etc/sipxpbx (httpd-sipxchange-common.conf.in, 
httpd-sipxchange-common-ssl.conf.in) but not the files themselves.

</description>
    <dc:creator>Joseph Attardi</dc:creator>
    <dc:date>2008-10-02T20:27:25</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12879">
    <title>fixing cgicc warnings</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12879</link>
    <description>I just submitted a patch to the lib/cgicc directory that I think will
eliminate the annoying new autoconf warning that appeared with the
upgrade to cgicc 3.2.7:

/usr/share/aclocal/cgicc.m4:31: warning: underquoted definition of CGICC_CHECK_CONFIG
/usr/share/aclocal/cgicc.m4:31:   run info '(automake)Extending aclocal'
/usr/share/aclocal/cgicc.m4:31:   or see http://sources.redhat.com/automake/automake.html#Extending-aclocal

I've attached the patch to this mail.  To apply, save it
to /tmp/cgicc.m4.patch and:

        cd /usr/share/aclocal
        sudo patch -p1 &lt; /tmp/cgicc.m4.patch

Mardy or Ken - if you would be so good as to rebuild the cgicc rpm; I
think the change I made to the spec file will apply this patch.

I'm trying to figure out where to send this upstream...

</description>
    <dc:creator>Scott Lawrence</dc:creator>
    <dc:date>2008-10-02T16:32:24</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12873">
    <title>HA DNS Setup</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12873</link>
    <description/>
    <dc:creator>Alex Samson</dc:creator>
    <dc:date>2008-10-02T14:47:40</dc:date>
  </item>
  <item rdf:about="http://comments.gmane.org/gmane.comp.voip.sipx.devel/12866">
    <title>64bit compilation issues for sipXregistry</title>
    <link>http://comments.gmane.org/gmane.comp.voip.sipx.devel/12866</link>
    <description/>
    <dc:creator>Ken Mahoney</dc:creator>
    <dc:date>2008-10-02T13:16:21</dc:date>
  </item>
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