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Giovanni Maruzzelli | 1 Dec 2010 01:15
Subject: Re: : Skypopen error

Re: : Skypopen error

Jian,

you're very very bizarre :)

Who is that is sending to you DTMF "D"????

First time I see this.

Normally only 0-9 * and # are used.

Good to know, no need to do the other debug log.

Skype client does not accept DTMF "A-D"

I'll fix that in the code, no problem.

Just for my curiosity: how is happening someone is sending you the "D" DTMF?

-giovanni

On Wed, Dec 1, 2010 at 12:54 AM, Jian Ren <renjian@...> wrote:
> Hi Giovanni,
> I got it happened with loglevel set to 9 on CentOS and created an issue as
> below, switched to Ubuntu, will add the Ubuntu log once it happened. Sorry
> that I can only do it one by one.
> http://jira.freeswitch.org/browse/FS-2891
> Regards!
> Jian
>
> On Fri, Nov 19, 2010 at 2:15 PM, Giovanni Maruzzelli <gmaruzz@...>
(Continue reading)

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Troy Anderson | 1 Dec 2010 05:42

DynamicRTP-Type-98

I noticed a problem with calls originating from the mod_callcenter.  The order of the codecs in the Media Description portion of the SDP is different for calls from mod_callcenter than when dialing the extension from another extension.  Specifically, ITU-T G.711 PCMU is coming first on calls that work, and DynamicRTP-Type-98 is coming first on those that don't.  A Polycom 450 is returning a 488 Not Acceptable Here message on the callcenter calls, and an Aastra 6731i has extremely choppy audio.  If I tell the Aastra to only allow G722, all works as expected.

I have my codecs set up in FS as global_codec_prefs=G7221 <at> 32000h,G7221 <at> 16000h,G722,PCMU,PCMA,GSM

Can anyone explain what DynamicRTP-Type-98 is and why it would be in the SDP when I'm not specifying it in the codec list?



Here's what the Media Description portion looks like when calling extension to extension:

            Media Description, name and address (m): audio 10140 RTP/AVP 0 98 99 9 8 3 101 13
                Media Type: audio
                Media Port: 10140
                Media Protocol: RTP/AVP
                Media Format: ITU-T G.711 PCMU
                Media Format: DynamicRTP-Type-98
                Media Format: DynamicRTP-Type-99
                Media Format: ITU-T G.722
                Media Format: ITU-T G.711 PCMA
                Media Format: GSM 06.10
                Media Format: DynamicRTP-Type-101
                Media Format: Comfort noise



Here's what the Media Description portion looks like when mod_callcenter initiates the call:

            Media Description, name and address (m): audio 12606 RTP/AVP 98 99 9 0 8 3 101 13
                Media Type: audio
                Media Port: 12606
                Media Protocol: RTP/AVP
                Media Format: DynamicRTP-Type-98
                Media Format: DynamicRTP-Type-99
                Media Format: ITU-T G.722
                Media Format: ITU-T G.711 PCMU
                Media Format: ITU-T G.711 PCMA
                Media Format: GSM 06.10
                Media Format: DynamicRTP-Type-101
                Media Format: Comfort noise


Thanks!

-Troy

<div>I noticed a problem with calls originating from the mod_callcenter. &nbsp;The order of the codecs in the Media Description portion of the SDP is different for calls from mod_callcenter than when dialing the extension from another extension. &nbsp;Specifically,&nbsp;ITU-T G.711 PCMU is coming first on calls that work, and DynamicRTP-Type-98 is coming first on those that don't. &nbsp;A Polycom 450 is returning a 488 Not Acceptable Here message on the callcenter calls, and an Aastra 6731i has extremely choppy audio. &nbsp;If I tell the Aastra to only allow G722, all works as expected.<div><br></div>
<div>I have my codecs set up in FS as&nbsp;<span class="Apple-style-span">global_codec_prefs=G7221 <at> 32000h,G7221 <at> 16000h,G722,PCMU,PCMA,GSM</span>
</div>
<div><span class="Apple-style-span"><br></span></div>
<div><span class="Apple-style-span">Can anyone explain what&nbsp;DynamicRTP-Type-98 is and why it would be in the SDP when I'm not specifying it in the codec list?</span></div>
<div><span class="Apple-style-span"><br></span></div>
<div><span class="Apple-style-span"><br></span></div>
<div><span class="Apple-style-span"><br></span></div>
<div><span class="Apple-style-span">Here's what the Media Description portion looks like when calling extension to extension:</span></div>
<div><span class="Apple-style-span"><br></span></div>
<div><span class="Apple-style-span"><div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Description, name and address (m): audio 10140 RTP/AVP 0 98 99 9 8 3 101 13</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Type: audio</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Port: 10140</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Protocol: RTP/AVP</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: ITU-T G.711 PCMU</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: DynamicRTP-Type-98</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: DynamicRTP-Type-99</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: ITU-T G.722</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: ITU-T G.711 PCMA</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: GSM 06.10</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: DynamicRTP-Type-101</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: Comfort noise</div>
<div><br></div>
<div><br></div>
<div><br></div>
<div>Here's what the Media Description portion looks like when mod_callcenter initiates the call:</div>
<div><br></div>
<div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Description, name and address (m): audio 12606 RTP/AVP 98 99 9 0 8 3 101 13</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Type: audio</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Port: 12606</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Protocol: RTP/AVP</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: DynamicRTP-Type-98</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: DynamicRTP-Type-99</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: ITU-T G.722</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: ITU-T G.711 PCMU</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: ITU-T G.711 PCMA</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: GSM 06.10</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: DynamicRTP-Type-101</div>
<div>&nbsp;&nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp; &nbsp;Media Format: Comfort noise</div>
</div>
<div><br></div></span></div>
<div><span class="Apple-style-span"><br></span></div>
<div><span class="Apple-style-span">Thanks!</span></div>
<div><span class="Apple-style-span"><br></span></div>
<div><span class="Apple-style-span">-Troy</span></div>
<div><span class="Apple-style-span"><br></span></div>
</div>
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Brian West | 1 Dec 2010 06:09
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Re: DynamicRTP-Type-98

Its because you are believing what your tool is telling you.. 98 is there because you are allowing it...
G7221 <at> 32000h,G7221 <at> 16000h are 98 and 99 in the dynamic range.  This is a bug in the polycom you'll neen to
set verbose_sdp=true in vars.xml and reload.  The polycom requires you to list all codecs in the rtp map
even thou the sdp is valid they seem to not look at any codec that isn't in the map.

/b

On Nov 30, 2010, at 10:42 PM, Troy Anderson wrote:

> Can anyone explain what DynamicRTP-Type-98 is and why it would be in the SDP when I'm not specifying it in
the codec list?
> 
>
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Dmitry Bely | 1 Dec 2010 09:59
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Re: FsGui Windows build

On Tue, Nov 30, 2010 at 7:09 PM, Jeff Lenk <jeff@...> wrote:
>
> Look at the current build of FSComm for windows for an example. It currently
> builds fine in Git Head with VS2008

Well, finally I have it compiling and working (don't see how FSComm
could help). I'm no expert in qmake so I just fixed qmake-generated
makefiles manually to make Microsoft's nmake happy. Now CPU 100% load
problem is over.

- Dmitry Bely
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abubacker | 1 Dec 2010 11:23
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mod_fifo or mod_callcenter

Dear all,
      I want to use mod_fifo or mod_callcenter to perform queuing 
operation , but I dont want this application to
connect the agent and the customer I have the external script to do that 
but it should give me the dial string of the
agent where the customer is likely to  connect with.

Is this possible or can we do this using some work around ?
Thanks in advance !

--

-- 
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy
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korn | 1 Dec 2010 11:47
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Re: Cant compile last git on FreeBSD

Thanks Michael, I have made fresh checkout at 29.11, and there is no changes

> Is this a fresh checkout or no? If not, bootstrap.sh and configure again
> and then make all.
> -MC
>
> On Tue, Nov 23, 2010 at 4:04 AM, korn <kornev@...> wrote:
> > Hello, I,m trying to compile last git on FreeBSD 7.1 and receive this
> > error in
> > mod_spandsp:
> >
> > making all mod_spandsp
> > Creating mod_spandsp_la-mod_spandsp_fax.lo
> > Compiling mod_spandsp_fax.c ...
> > cc1: warnings being treated as errors
> > mod_spandsp_fax.c: In function 'configure_t38':
> > mod_spandsp_fax.c:721: warning: implicit declaration of function
> > 't38_set_fastest_image_data_rate'
> > mod_spandsp_fax.c: In function 't38_gateway_on_consume_media':
> > mod_spandsp_fax.c:1637: warning: implicit declaration of function
> > 't38_gateway_rx_fillin'
> > gmake[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1
> > gmake[3]: *** [mod_spandsp-all] Error 1
> > gmake[2]: *** [all-recursive] Error 1
> >
> >
> > configure and gmake in ./libs/spandsp are successful
> >
> > can anybody tell me, why I'm getting error
> >
> > WBR
> > Evgeny
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@...
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
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afshin afzali | 1 Dec 2010 11:50
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Re: mod_fifo or mod_callcenter

Abubacker,

I think you can use loopback endpoint in your dial-string to go through an extension. Look at the "Another example of On-hook Agent Login/Logout with loopback members"
in wiki.

-- afshin

On Wed, Dec 1, 2010 at 1:53 PM, abubacker <abubacker-Smb7ZHgQuQmWX3Rh8cA0QQ@public.gmane.org> wrote:
Dear all,
     I want to use mod_fifo or mod_callcenter to perform queuing
operation , but I dont want this application to
connect the agent and the customer I have the external script to do that
but it should give me the dial string of the
agent where the customer is likely to  connect with.

Is this possible or can we do this using some work around ?
Thanks in advance !

--
Best regards,
N.Abubacker ,
Associate system engineer ,
bk systems pvt ltd ,
Ph : 9144-43902701

Disclaimer: http://www.bksystems.co.in/email-policy


_______________________________________________
FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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<div>
<p>Abubacker,<br><br>I think you can use loopback endpoint in your dial-string to go through an extension. Look at the "<span class="mw-headline">Another example of On-hook Agent Login/Logout with loopback members</span>"<br>
in wiki.<br><br>-- afshin<br><br></p>
<div class="gmail_quote">On Wed, Dec 1, 2010 at 1:53 PM, abubacker <span dir="ltr">&lt;<a href="mailto:abubacker <at> bksystems.co.in" target="_blank">abubacker@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">Dear all,<br>
 &nbsp; &nbsp; &nbsp;I want to use mod_fifo or mod_callcenter to perform queuing<br>
operation , but I dont want this application to<br>
connect the agent and the customer I have the external script to do that<br>
but it should give me the dial string of the<br>
agent where the customer is likely to &nbsp;connect with.<br><br>
Is this possible or can we do this using some work around ?<br>
Thanks in advance !<br><br>
--<br>
Best regards,<br>
N.Abubacker ,<br>
Associate system engineer ,<br>
bk systems pvt ltd ,<br>
Ph : 9144-43902701<br><br>
Disclaimer: <a href="http://www.bksystems.co.in/email-policy" target="_blank">http://www.bksystems.co.in/email-policy</a><br><br><br>
_______________________________________________<br>
FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@..." target="_blank">FreeSWITCH-users@...</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</blockquote>
</div>
<br>
</div>
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Erik Dekkers | 1 Dec 2010 11:58
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Re: Cant compile last git on FreeBSD

Already tried http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD with first installing libtiff?

I've compiled yesterday on Freebsd 8.1 with no problems.

-----Oorspronkelijk bericht-----
Van: freeswitch-users-bounces@...
[mailto:freeswitch-users-bounces@...] Namens korn
Verzonden: woensdag 1 december 2010 11:47
Aan: FreeSWITCH Users Help
Onderwerp: Re: [Freeswitch-users] Cant compile last git on FreeBSD

Thanks Michael, I have made fresh checkout at 29.11, and there is no changes

> Is this a fresh checkout or no? If not, bootstrap.sh and configure 
> again and then make all.
> -MC
>
> On Tue, Nov 23, 2010 at 4:04 AM, korn <kornev@...> wrote:
> > Hello, I,m trying to compile last git on FreeBSD 7.1 and receive 
> > this error in
> > mod_spandsp:
> >
> > making all mod_spandsp
> > Creating mod_spandsp_la-mod_spandsp_fax.lo Compiling 
> > mod_spandsp_fax.c ...
> > cc1: warnings being treated as errors
> > mod_spandsp_fax.c: In function 'configure_t38':
> > mod_spandsp_fax.c:721: warning: implicit declaration of function 
> > 't38_set_fastest_image_data_rate'
> > mod_spandsp_fax.c: In function 't38_gateway_on_consume_media':
> > mod_spandsp_fax.c:1637: warning: implicit declaration of function 
> > 't38_gateway_rx_fillin'
> > gmake[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1
> > gmake[3]: *** [mod_spandsp-all] Error 1
> > gmake[2]: *** [all-recursive] Error 1
> >
> >
> > configure and gmake in ./libs/spandsp are successful
> >
> > can anybody tell me, why I'm getting error
> >
> > WBR
> > Evgeny
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users@...
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > sers
> > http://www.freeswitch.org

_______________________________________________
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korn | 1 Dec 2010 12:31
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Re: Cant compile last git on FreeBSD


server# ll /usr/local/lib/ |grep tiff
-rw-r--r--  1 root  wheel   405936 Jun 29  2009 libtiff.a
-rwxr-xr-x  1 root  wheel      837 Jun 29  2009 libtiff.la
lrwxr-xr-x  1 root  wheel       12 Jun 29  2009 libtiff.so -> libtiff.so.4
-rwxr-xr-x  1 root  wheel   372497 Jun 29  2009 libtiff.so.4
-rw-r--r--  1 root  wheel     6384 Jun 29  2009 libtiffxx.a
-rwxr-xr-x  1 root  wheel      873 Jun 29  2009 libtiffxx.la
lrwxr-xr-x  1 root  wheel       14 Jun 29  2009 libtiffxx.so -> libtiffxx.so.4
-rwxr-xr-x  1 root  wheel    11331 Jun 29  2009 libtiffxx.so.4
server#
server# ll /usr/local/include/ | grep tiff
-r--r--r--  1 root  wheel  33725 Jun 29  2009 tiff.h
-r--r--r--  1 root  wheel   2867 Jun 29  2009 tiffconf.h
-r--r--r--  1 root  wheel  19711 Jun 29  2009 tiffio.h
-r--r--r--  1 root  wheel   1610 Jun 29  2009 tiffio.hxx
-r--r--r--  1 root  wheel    410 Jun 29  2009 tiffvers.h

and configure and gmake in ./libs/spandsp are successful

> Already tried http://wiki.freeswitch.org/wiki/Installation_Guide#FreeBSD
> with first installing libtiff?
>
> I've compiled yesterday on Freebsd 8.1 with no problems.
>
> -----Oorspronkelijk bericht-----
> Van: freeswitch-users-bounces@...
> [mailto:freeswitch-users-bounces@...] Namens korn
> Verzonden: woensdag 1 december 2010 11:47
> Aan: FreeSWITCH Users Help
> Onderwerp: Re: [Freeswitch-users] Cant compile last git on FreeBSD
>
> Thanks Michael, I have made fresh checkout at 29.11, and there is no
> changes
>
> > Is this a fresh checkout or no? If not, bootstrap.sh and configure
> > again and then make all.
> > -MC
> >
> > On Tue, Nov 23, 2010 at 4:04 AM, korn <kornev@...> wrote:
> > > Hello, I,m trying to compile last git on FreeBSD 7.1 and receive
> > > this error in
> > > mod_spandsp:
> > >
> > > making all mod_spandsp
> > > Creating mod_spandsp_la-mod_spandsp_fax.lo Compiling
> > > mod_spandsp_fax.c ...
> > > cc1: warnings being treated as errors
> > > mod_spandsp_fax.c: In function 'configure_t38':
> > > mod_spandsp_fax.c:721: warning: implicit declaration of function
> > > 't38_set_fastest_image_data_rate'
> > > mod_spandsp_fax.c: In function 't38_gateway_on_consume_media':
> > > mod_spandsp_fax.c:1637: warning: implicit declaration of function
> > > 't38_gateway_rx_fillin'
> > > gmake[4]: *** [mod_spandsp_la-mod_spandsp_fax.lo] Error 1
> > > gmake[3]: *** [mod_spandsp-all] Error 1
> > > gmake[2]: *** [all-recursive] Error 1
> > >
> > >
> > > configure and gmake in ./libs/spandsp are successful
> > >
> > > can anybody tell me, why I'm getting error
> > >
> > > WBR
> > > Evgeny
> > >
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users@...
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u
> > > sers
> > > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users@...
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users@...
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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Seven Du | 1 Dec 2010 14:51
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dead call records

Hi,

I found a wired problem where left dead call records.

originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park
originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park
uuid_bridge 1 2
uuid_transfer 1 -both park inline
hupall

show channels # show nothing
show calls # the record still there, why the uuid is not 1 and 2 but a
long uuid str?

but there's no problem if not use origination_uuid.

Also, there are no problems if no uuid_transfer.

originate {origination_uuid=1,hangup_after_bridge=false}user/1000 &park
originate {origination_uuid=2,hangup_after_bridge=false}user/1001 &park
uuid_bridge 1 2
hupall

I'm on last git, anyone can help take a look?

http://pastebin.freeswitch.org/14681

Thanks.

--

-- 
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj:  http://www.freeswitch.org.cn

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Gmane