From jan.t.ohlsen at gmail.com  Sun Dec  2 10:54:09 2012
From: jan.t.ohlsen at gmail.com (Jan Ohlsen)
Date: Sun, 2 Dec 2012 08:54:09 +0100
Subject: [Freeswitch-dev] mod_lua: LuaJIT ?
Message-ID: <CAMDWk-msMq68QjxVDfGtgb6SXbA07_Yw+F+1DYOdj13iprceJQ@mail.gmail.com>

Seems like LuaJIT could offer a "drop-in" performance boost for mod_lua .. ?
http://luajit.org/performance_x86.html

"LuaJIT is API-compatible with Lua 5.1. If you've already embedded Lua into
your application, you probably don't need to do anything to switch to
LuaJIT, except link with a different library ..."
http://luajit.org/install.html#embed


Regards,

  JT
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From sparklezou at 163.com  Sun Dec  2 12:03:24 2012
From: sparklezou at 163.com (sparklezou)
Date: Sun, 2 Dec 2012 17:03:24 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display problem
	during calling
In-Reply-To: <35a86017.4737.13b51737748.Coremail.sparklezou@163.com>
References: <5c994372.3147.13b4f55d827.Coremail.sparklezou@163.com><79e311e1.6599.13b504fd0fa.Coremail.sparklezou@163.com><35a86017.4737.13b51737748.Coremail.sparklezou@163.com>
Message-ID: <4000228c.a63.13b5adccbd8.Coremail.sparklezou@163.com>

Hi Develop team,

In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:

Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.


FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.

Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.

Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.

Then in the following whole process, there will be display name both for "from" & "to".

How about your idea?

Thanks in advance!

2012-12-02



sparklezou



????sparklezou
?????2012-11-30 21:13
???Re: Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to 

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
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From anthony.minessale at gmail.com  Mon Dec  3 03:05:44 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 2 Dec 2012 18:05:44 -0600
Subject: [Freeswitch-dev] mod_lua: LuaJIT ?
In-Reply-To: <CAMDWk-msMq68QjxVDfGtgb6SXbA07_Yw+F+1DYOdj13iprceJQ@mail.gmail.com>
References: <CAMDWk-msMq68QjxVDfGtgb6SXbA07_Yw+F+1DYOdj13iprceJQ@mail.gmail.com>
Message-ID: <CAKbxfG_FgAvNQ5X7Cf9wju3OFe=3M85ocN89PpxwXH5aOhEvrw@mail.gmail.com>

Updating lua is going to be a bit of a can of worms regarding swig and some
manual hacks we have to do with the generated files...

We will get there but it may be a while....
On Dec 2, 2012 2:26 AM, "Jan Ohlsen" <jan.t.ohlsen at gmail.com> wrote:

>
> Seems like LuaJIT could offer a "drop-in" performance boost for mod_lua ..
> ?
> http://luajit.org/performance_x86.html
>
> "LuaJIT is API-compatible with Lua 5.1. If you've already embedded Lua
> into your application, you probably don't need to do anything to switch to
> LuaJIT, except link with a different library ..."
> http://luajit.org/install.html#embed
>
>
> Regards,
>
>   JT
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From sparklezou at 163.com  Mon Dec  3 04:51:08 2012
From: sparklezou at 163.com (sparklezou)
Date: Mon, 3 Dec 2012 09:51:08 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display
	problem	during calling
In-Reply-To: <4000228c.a63.13b5adccbd8.Coremail.sparklezou@163.com>
References: <4000228c.a63.13b5adccbd8.Coremail.sparklezou@163.com>
Message-ID: <657d209d.1b68.13b5e7768e9.Coremail.sparklezou@163.com>

Hi Brian,

It's the default Dialplan.

I have tried the fresh last version also. The same result. It should NOT be configuration problem.

You could review the wireshark log.

Thanks!

2012-12-03



sparklezou



????sparklezou
?????2012-12-02 17:03
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:

Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.


FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.

Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.

Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.

Then in the following whole process, there will be display name both for "from" & "to".

How about your idea?

Thanks in advance!

2012-12-02



sparklezou



????sparklezou
?????2012-11-30 21:13
???Re: Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to 

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
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From sparklezou at 163.com  Mon Dec  3 13:30:37 2012
From: sparklezou at 163.com (sparklezou)
Date: Mon, 3 Dec 2012 18:30:37 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display
	problem	during calling
In-Reply-To: <657d209d.1b68.13b5e7768e9.Coremail.sparklezou@163.com>
References: <4000228c.a63.13b5adccbd8.Coremail.sparklezou@163.com><657d209d.1b68.13b5e7768e9.Coremail.sparklezou@163.com>
Message-ID: <240b9b25.9346.13b60538011.Coremail.sparklezou@163.com>

Hi Brian,

I'm trying to find where the session start.

Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.

My idea, when get the INVITE SIP message from the sofia stack. check the called number detail info in FS. add the dispaly name in the "to" field of the following SIP message.

It should finish before the new channel.

where could add such code? 

Thanks!


2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel sofia/internal/1001 at 172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1001 at 172.0.0.10) Running State Change CS_NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1001 at 172.0.0.10) State NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel sofia/internal/1001 at 172.0.0.10 entering state [received][100]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:

2012-12-03



sparklezou



????sparklezou
?????2012-12-03 09:53
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"brian"<brian at freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Brian,

It's the default Dialplan.

I have tried the fresh last version also. The same result. It should NOT be configuration problem.

You could review the wireshark log.

Thanks!

2012-12-03



sparklezou



????sparklezou
?????2012-12-02 17:03
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:

Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.


FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.

Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.

Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.

Then in the following whole process, there will be display name both for "from" & "to".

How about your idea?

Thanks in advance!

2012-12-02



sparklezou



????sparklezou
?????2012-11-30 21:13
???Re: Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to 

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
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From mike at jerris.com  Mon Dec  3 18:20:41 2012
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Dec 2012 09:20:41 -0600
Subject: [Freeswitch-dev] [Freeswitch-users] About the display problem
	during calling
In-Reply-To: <240b9b25.9346.13b60538011.Coremail.sparklezou@163.com>
References: <4000228c.a63.13b5adccbd8.Coremail.sparklezou@163.com><657d209d.1b68.13b5e7768e9.Coremail.sparklezou@163.com>
	<240b9b25.9346.13b60538011.Coremail.sparklezou@163.com>
Message-ID: <65B9D5C2-3A8E-441C-ABFD-C6BB57FDA896@jerris.com>

All of this is handled in mod_sofia, look for a function with i_invite in the name.

Mike

On Dec 3, 2012, at 4:30 AM, sparklezou <sparklezou at 163.com> wrote:

> Hi Brian,
>  
> I'm trying to find where the session start.
>  
> Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.
>  
> My idea, when get the INVITE SIP message from the sofia stack. check the called number detail info in FS. add the dispaly name in the "to" field of the following SIP message.
>  
> It should finish before the new channel.
>  
> where could add such code?
>  
> Thanks!
>  
>  
> 2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel sofia/internal/1001 at 172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc]
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1001 at 172.0.0.10) Running State Change CS_NEW
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1001 at 172.0.0.10) State NEW
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
> 2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc
> 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc
> 2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
> 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel sofia/internal/1001 at 172.0.0.10 entering state [received][100]
> 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:
>  
> 2012-12-03
> sparklezou
> ????sparklezou
> ?????2012-12-03 09:53
> ???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
> ????"brian"<brian at freeswitch.org>
> ???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
>  
> Hi Brian,
>  
> It's the default Dialplan.
>  
> I have tried the fresh last version also. The same result. It should NOT be configuration problem.
>  
> You could review the wireshark log.
>  
> Thanks!
>  
> 2012-12-03
> sparklezou
> ????sparklezou
> ?????2012-12-02 17:03
> ???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
> ????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> ???
>  
> Hi Develop team,
>  
> In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:
>  
> Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.
>  
>  
> FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.
>  
> Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.
>  
> Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.
>  
> Then in the following whole process, there will be display name both for "from" & "to".
>  
> How about your idea?
>  
> Thanks in advance!
>  
> 2012-12-02
> sparklezou
> ????sparklezou
> ?????2012-11-30 21:13
> ???Re: Re: [Freeswitch-users] About the display problem during calling
> ????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> ???
>  
> Hi Develop team,
>  
> Just simlify this question.
>  
> In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.
>  
> Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.
>  
> Could you please let me know where is the source code for such process? I also would like to review it.
>  
> Thanks!
>  
> 2012-11-30
> sparklezou
> ????sparklezou
> ?????2012-11-30 15:55
> ???Re: [Freeswitch-users] About the display problem during calling
> ????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> ???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>
>  
> Hi All,
>  
> Before, I saved the user name in the phone. So it will show the name when ringing.
>  
> From the FreeSwitch side, there is NO difference.
>  
> @ Develop Team,
>  
> For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".
>  
> Usually the phone will update the display name from the sip response message.
>  
> So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.
>  
> Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "
>  
> If it could be updated, then the called name could be displayed on the caller phone.
>  
> Thanks!
>  
> 2012-11-30
> sparklezou
> ????sparklezou
> ?????2012-11-30 11:19
> ???[Freeswitch-users] About the display problem during calling
> ????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> ???
>  
> Hi All,
>  
> I facing a strange problem.
>  
> Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.
>  
> But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.
>  
> I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.
>  
> I checked the sip info.
>  
> The "Outbound Call" is in the 200 SDP message. the called is set to
>  
> ""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "
>  
> Not the correct called name.
>  
> Is there anything wrong?
>  
> Please help me.
>  
> Thanks!
>  
>  
> 2012-11-30
> sparklezou
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From mike at jerris.com  Mon Dec  3 18:25:09 2012
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Dec 2012 09:25:09 -0600
Subject: [Freeswitch-dev] mod_lua: LuaJIT ?
In-Reply-To: <CAKbxfG_FgAvNQ5X7Cf9wju3OFe=3M85ocN89PpxwXH5aOhEvrw@mail.gmail.com>
References: <CAMDWk-msMq68QjxVDfGtgb6SXbA07_Yw+F+1DYOdj13iprceJQ@mail.gmail.com>
	<CAKbxfG_FgAvNQ5X7Cf9wju3OFe=3M85ocN89PpxwXH5aOhEvrw@mail.gmail.com>
Message-ID: <0ECC97E4-F4E6-4711-969D-4C69355E370D@jerris.com>

If its really a drop in replacement and doesn't require any changes to the swig code, then we might get away with a bit less work, but it would need swapping out the lib, and related build changes.

Mike

On Dec 2, 2012, at 6:05 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Updating lua is going to be a bit of a can of worms regarding swig and some manual hacks we have to do with the generated files...
> 
> We will get there but it may be a while....
> 
> On Dec 2, 2012 2:26 AM, "Jan Ohlsen" <jan.t.ohlsen at gmail.com> wrote:
> 
> Seems like LuaJIT could offer a "drop-in" performance boost for mod_lua .. ?
> http://luajit.org/performance_x86.html
> 
> "LuaJIT is API-compatible with Lua 5.1. If you've already embedded Lua into your application, you probably don't need to do anything to switch to LuaJIT, except link with a different library ..."
> http://luajit.org/install.html#embed
> 
> 
> Regards,
> 
>   JT
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From sparklezou at 163.com  Mon Dec  3 18:38:54 2012
From: sparklezou at 163.com (sparklezou)
Date: Mon, 3 Dec 2012 23:38:54 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display
	problem	during calling
In-Reply-To: <65B9D5C2-3A8E-441C-ABFD-C6BB57FDA896@jerris.com>
References: <65B9D5C2-3A8E-441C-ABFD-C6BB57FDA896@jerris.com>
Message-ID: <b07ecce.5acd.13b616d67b3.Coremail.sparklezou@163.com>

Hi Mike,

Thank!

In which functin should be the right place to add the Name for the "to" field? 

How about your idea?

2012-12-03



sparklezou



????Michael Jerris
?????2012-12-03 23:20
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

All of this is handled in mod_sofia, look for a function with i_invite in the name.


Mike


On Dec 3, 2012, at 4:30 AM, sparklezou <sparklezou at 163.com> wrote:


Hi Brian,

I'm trying to find where the session start.

Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.

My idea, when get the INVITE SIP message from the sofia stack. check the called number detail info in FS. add the dispaly name in the "to" field of the following SIP message.

It should finish before the new channel.

where could add such code?

Thanks!


2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel sofia/internal/1001 at 172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1001 at 172.0.0.10) Running State Change CS_NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1001 at 172.0.0.10) State NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel sofia/internal/1001 at 172.0.0.10 entering state [received][100]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:

2012-12-03



sparklezou



????sparklezou
?????2012-12-03 09:53
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"brian"<brian at freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Brian,

It's the default Dialplan.

I have tried the fresh last version also. The same result. It should NOT be configuration problem.

You could review the wireshark log.

Thanks!

2012-12-03



sparklezou



????sparklezou
?????2012-12-02 17:03
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:

Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.


FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.

Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.

Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.

Then in the following whole process, there will be display name both for "from" & "to".

How about your idea?

Thanks in advance!

2012-12-02



sparklezou



????sparklezou
?????2012-11-30 21:13
???Re: Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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From marketing at cluecon.com  Mon Dec  3 23:04:08 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 3 Dec 2012 12:04:08 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxUssdCHM-NXL9pQyAZixU7=0mA5c-a=fE9kS=iz_7hApA@mail.gmail.com>

Happy December to everyone!

Last week was painful for many of us as we were dealing with a sustained
DDoS attack on most of our infrastructure. Kudos to the guys for working
through it. It seems the worst is over and we can get back to the business
at hand: doing FreeSWITCH stuff. :)

In spite of the drama last week we did have a conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_11_28>and we
released 1.2.5.2! We discussed mostly the details of the DDoS we
experienced and how the community can assist in the future so that we can
mitigate the effects of such an occurrence. With the community's help we
will be more resistant to the effects of any future attacks. We appreciate
the outpouring of support we received from everyone.

This week <http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05> we will go
back to discussing FreeSWITCH. We are still finalizing future guests so
this week we'll do another installment of tips and tricks from the
FreeSWITCH community. Among other things I will be showing how Chris Rienzo
(IRC: crienzo) and I used the source this weekend to figure out what the
XML preprocessor can do and get the wiki updated. I'll then show a simple
example of the always-present-but-previously-
undocumented command can do. As an added bonus we'll have an update on the
ClueCon <http://www.cluecon.com> 2012 videos!

Thanks and have a great week.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From dujinfang at gmail.com  Tue Dec  4 03:56:09 2012
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 4 Dec 2012 08:56:09 +0800
Subject: [Freeswitch-dev] Conference CDRs via http
In-Reply-To: <CAKbxfG_VxhyzqH3GWmN35khJ5WOY7S0GK4CbpvOUb5jZUt9T_g@mail.gmail.com>
References: <4fd69dad04abd6.16863237@wp.pl>
	<CAKbxfG_VxhyzqH3GWmN35khJ5WOY7S0GK4CbpvOUb5jZUt9T_g@mail.gmail.com>
Message-ID: <4E225564E2EA4577B7356C390E21B89A@gmail.com>

Any update on this? 

How about just fire a SWITCH_EVENT_CDR  with raw event headers so mod like mod_cdr_pg_csv can listen to it and write to db without parsing back the XML?

Thanks.

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Tuesday, June 12, 2012 at 10:31 AM, Anthony Minessale wrote:

> B is the right choice.
> The size of the body has no limit. 
> How bout, if a certian header exists in the event with a local file path, then the the mod opens the referenced file for delivery; Otherwise it uses the body if it exists.
> On Jun 11, 2012 8:39 PM, "Mariusz Czulada" <manieq at wp.eu (mailto:manieq at wp.eu)> wrote:
> > Hi all,
> > 
> > I was thinking about sending CDRs via HTTP in a same or similar way mod_xml_cdr does. I consider implementing this (unless someone else is working on it) but first wanted to discuss with you the best approach.
> > 
> > 
> > A. All in mod_conference
> > 
> > This would require to copy many fragments of code from mod_xml_cdr into mod_conference. Also same configuration parameters used in xml_cdr must be processed and used for sending data. The advantage for this solution is that everything related to this mechanism is included in this module. Drawback: if something in mod_xml_cdr requires fixing or extending, probably the same changes should by applied in related parts of mod_conference.
> > 
> > 
> > B. mod_conference builds, mod_xml_cdr sends data
> > 
> > For this solution I'll give more details.
> > 1. New event type should be added (like SWITCH_EVENT_CDR)
> > 2. When a module (in this case: mod_conference) wants to store CDR via HTTP it must fire an event of that type and:
> > - "Event-Subclass" set (like "conference")
> > - Only common headers are needed, plus...
> > - "Content-Type" and "Content-Length" must be set
> > - CDR data must be build as XML in a module and added as an event content.
> > 3. mod_xml_cdr will listen to this event type.
> > 4. For each event subclass which mod_xml_cdr must must react, configuration file will contain a set of params same as for generic channel CDRs.
> > 5. If subclass matches configuration, mod_xml_cdr reads data from event content and sends them according to configuration.
> > 
> > Changes in mod_conference:
> > - one new parameter for each profile (like "cdr-via-event=yes|no")
> > - if "yes" then xml must be build even if "cdr-log-dir" is unset
> > - if "yes", then an event must be fired as described above
> > 
> > Changes in mod_xml_cdr:
> > - extra parameters from configuration to be parsed (like '<settings subclass="conference">....</settings>')
> > - bind to SWITCH_EVENT_CDR
> > - if 'Event-Subclass' matches configuration, a content of the event will be sent via HTTP (probably most of 'my_on_report' routine bellow 'try to post it to the web server' comment will be reused)
> > 
> > A [small] drawback is that it makes an indirect module dependency, but we already have such situations (like mod_shout needed to record a conference in mp3).
> > 
> > Advantages are:
> > - one can create an external tool for handling this type of event (to store it directly in db or send it with other protocols)
> > - this mechanism can be easily reused in other modules if needed; maybe in mod_callcenter, maybe in other components. No further changes in mod_xml_cdr should be needed.
> > 
> > What is unknown to me is a maximum size of event content. Conference CDR XMLs can be quite big - will it be a problem to send it this way?
> > 
> > 
> > I think the second solution is better and more universal but I'd like to hear your opinions about this case.
> > 
> > Regards,
> > 
> > Mariusz
> > 
> > 
> > 
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> > 
> > Join Us At ClueCon - Aug 7-9, 2012
> > 
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org)
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org (mailto:FreeSWITCH-dev at lists.freeswitch.org)
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
> 
> 


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From anthony.minessale at gmail.com  Tue Dec  4 04:00:15 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 Dec 2012 19:00:15 -0600
Subject: [Freeswitch-dev] Conference CDRs via http
In-Reply-To: <4E225564E2EA4577B7356C390E21B89A@gmail.com>
References: <4fd69dad04abd6.16863237@wp.pl>
	<CAKbxfG_VxhyzqH3GWmN35khJ5WOY7S0GK4CbpvOUb5jZUt9T_g@mail.gmail.com>
	<4E225564E2EA4577B7356C390E21B89A@gmail.com>
Message-ID: <CAKbxfG8RfSY1EqUZkERcAb-QD2sfHAhRgGSHYMeZzVJnVV-baw@mail.gmail.com>

I have no had time to do it but if you want to start a patch I can work on
getting it in.



On Mon, Dec 3, 2012 at 6:56 PM, Seven Du <dujinfang at gmail.com> wrote:

>  Any update on this?
>
> How about just fire a SWITCH_EVENT_CDR  with raw event headers so mod like
> mod_cdr_pg_csv can listen to it and write to db without parsing back the
> XML?
>
> Thanks.
>
> --
> Seven Du
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Tuesday, June 12, 2012 at 10:31 AM, Anthony Minessale wrote:
>
> B is the right choice.
> The size of the body has no limit.
>
> How bout, if a certian header exists in the event with a local file path,
> then the the mod opens the referenced file for delivery; Otherwise it uses
> the body if it exists.
>  On Jun 11, 2012 8:39 PM, "Mariusz Czulada" <manieq at wp.eu> wrote:
>
> Hi all,
>
> I was thinking about sending CDRs via HTTP in a same or similar way
> mod_xml_cdr does. I consider implementing this (unless someone else is
> working on it) but first wanted to discuss with you the best approach.
>
>
> A. All in mod_conference
>
> This would require to copy many fragments of code from mod_xml_cdr into
> mod_conference. Also same configuration parameters used in xml_cdr must be
> processed and used for sending data. The advantage for this solution is
> that everything related to this mechanism is included in this module.
> Drawback: if something in mod_xml_cdr requires fixing or extending,
> probably the same changes should by applied in related parts of
> mod_conference.
>
>
> B. mod_conference builds, mod_xml_cdr sends data
>
> For this solution I'll give more details.
> 1. New event type should be added (like SWITCH_EVENT_CDR)
> 2. When a module (in this case: mod_conference) wants to store CDR via
> HTTP it must fire an event of that type and:
> - "Event-Subclass" set (like "conference")
> - Only common headers are needed, plus...
> - "Content-Type" and "Content-Length" must be set
> - CDR data must be build as XML in a module and added as an event content.
> 3. mod_xml_cdr will listen to this event type.
> 4. For each event subclass which mod_xml_cdr must must react,
> configuration file will contain a set of params same as for generic channel
> CDRs.
> 5. If subclass matches configuration, mod_xml_cdr reads data from event
> content and sends them according to configuration.
>
> Changes in mod_conference:
> - one new parameter for each profile (like "cdr-via-event=yes|no")
> - if "yes" then xml must be build even if "cdr-log-dir" is unset
> - if "yes", then an event must be fired as described above
>
> Changes in mod_xml_cdr:
> - extra parameters from configuration to be parsed (like '<settings
> subclass="conference">....</settings>')
> - bind to SWITCH_EVENT_CDR
> - if 'Event-Subclass' matches configuration, a content of the event will
> be sent via HTTP (probably most of 'my_on_report' routine bellow 'try to
> post it to the web server' comment will be reused)
>
> A [small] drawback is that it makes an indirect module dependency, but we
> already have such situations (like mod_shout needed to record a conference
> in mp3).
>
> Advantages are:
> - one can create an external tool for handling this type of event (to
> store it directly in db or send it with other protocols)
> - this mechanism can be easily reused in other modules if needed; maybe in
> mod_callcenter, maybe in other components. No further changes in
> mod_xml_cdr should be needed.
>
> What is unknown to me is a maximum size of event content. Conference CDR
> XMLs can be quite big - will it be a problem to send it this way?
>
>
> I think the second solution is better and more universal but I'd like to
> hear your opinions about this case.
>
> Regards,
>
> Mariusz
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
-------------- next part --------------
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From sparklezou at 163.com  Tue Dec  4 09:43:08 2012
From: sparklezou at 163.com (sparklezou)
Date: Tue, 4 Dec 2012 14:43:08 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display
	problem	during calling
In-Reply-To: <b07ecce.5acd.13b616d67b3.Coremail.sparklezou@163.com>
References: <65B9D5C2-3A8E-441C-ABFD-C6BB57FDA896@jerris.com><b07ecce.5acd.13b616d67b3.Coremail.sparklezou@163.com>
Message-ID: <6e81da06.56bc.13b64a9644b.Coremail.sparklezou@163.com>

Hi Mike & Brian,

I  find the way to correct the final 200 SDP response message.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

In the switch_ivr_originate.c file,  in the following code.
    
      new_profile->callee_id_name = switch_core_strdup(new_profile->pool, "Outbound Call");

Before this sentence, there should be one function to find the display name from the number. Then replace the default "Outbound Call".

Is there any function to find the name from the user xml files?

In this way, just correct the final dispaly issue.

And during the call, I still NOT find the corret location to add the dispaly name in to the sip "to" field.

But I believe it should NOT add in the sofia sip stack side. It should add at the uplay application invote function side.

I think, it should be in switch_channel.c switch_ivr_originate.c switch_event.c files.

Please check and consider it.

Thanks!


2012-12-04



sparklezou



????sparklezou
?????2012-12-03 23:40
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Mike,

Thank!

In which functin should be the right place to add the Name for the "to" field? 

How about your idea?

2012-12-03



sparklezou



????Michael Jerris
?????2012-12-03 23:20
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

All of this is handled in mod_sofia, look for a function with i_invite in the name. 


Mike


On Dec 3, 2012, at 4:30 AM, sparklezou <sparklezou at 163.com> wrote:


Hi Brian,

I'm trying to find where the session start.

Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.

My idea, when get the INVITE SIP message from the sofia stack. check the called number detail info in FS. add the dispaly name in the "to" field of the following SIP message.

It should finish before the new channel.

where could add such code?

Thanks!


2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel sofia/internal/1001 at 172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1001 at 172.0.0.10) Running State Change CS_NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1001 at 172.0.0.10) State NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel sofia/internal/1001 at 172.0.0.10 entering state [received][100]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:

2012-12-03



sparklezou



????sparklezou
?????2012-12-03 09:53
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"brian"<brian at freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Brian,

It's the default Dialplan.

I have tried the fresh last version also. The same result. It should NOT be configuration problem.

You could review the wireshark log.

Thanks!

2012-12-03



sparklezou



????sparklezou
?????2012-12-02 17:03
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:

Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.


FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.

Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.

Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.

Then in the following whole process, there will be display name both for "from" & "to".

How about your idea?

Thanks in advance!

2012-12-02



sparklezou



????sparklezou
?????2012-11-30 21:13
???Re: Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
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From sparklezou at 163.com  Tue Dec  4 13:57:15 2012
From: sparklezou at 163.com (sparklezou)
Date: Tue, 4 Dec 2012 18:57:15 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display
	problem	during calling
In-Reply-To: <6e81da06.56bc.13b64a9644b.Coremail.sparklezou@163.com>
References: <65B9D5C2-3A8E-441C-ABFD-C6BB57FDA896@jerris.com><b07ecce.5acd.13b616d67b3.Coremail.sparklezou@163.com><6e81da06.56bc.13b64a9644b.Coremail.sparklezou@163.com>
Message-ID: <58dc18bb.8e37.13b65920a8f.Coremail.sparklezou@163.com>

Hi Mike,

I think, the search number/Name function could refer to "list_users_function" function in mod_commands.c

How about your idea?

Thanks!

2012-12-04



sparklezou



????sparklezou
?????2012-12-04 14:45
???Re: Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Mike & Brian,

I  find the way to correct the final 200 SDP response message.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

In the switch_ivr_originate.c file,  in the following code.
    
      new_profile->callee_id_name = switch_core_strdup(new_profile->pool, "Outbound Call");

Before this sentence, there should be one function to find the display name from the number. Then replace the default "Outbound Call".

Is there any function to find the name from the user xml files?

In this way, just correct the final dispaly issue.

And during the call, I still NOT find the corret location to add the dispaly name in to the sip "to" field.

But I believe it should NOT add in the sofia sip stack side. It should add at the uplay application invote function side.

I think, it should be in switch_channel.c switch_ivr_originate.c switch_event.c files.

Please check and consider it.

Thanks!


2012-12-04



sparklezou



????sparklezou
?????2012-12-03 23:40
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Mike,

Thank!

In which functin should be the right place to add the Name for the "to" field? 

How about your idea?

2012-12-03



sparklezou



????Michael Jerris
?????2012-12-03 23:20
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

All of this is handled in mod_sofia, look for a function with i_invite in the name. 


Mike


On Dec 3, 2012, at 4:30 AM, sparklezou <sparklezou at 163.com> wrote:


Hi Brian,

I'm trying to find where the session start.

Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.

My idea, when get the INVITE SIP message from the sofia stack. check the called number detail info in FS. add the dispaly name in the "to" field of the following SIP message.

It should finish before the new channel.

where could add such code?

Thanks!


2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel sofia/internal/1001 at 172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1001 at 172.0.0.10) Running State Change CS_NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1001 at 172.0.0.10) State NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel sofia/internal/1001 at 172.0.0.10 entering state [received][100]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:

2012-12-03



sparklezou



????sparklezou
?????2012-12-03 09:53
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"brian"<brian at freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Brian,

It's the default Dialplan.

I have tried the fresh last version also. The same result. It should NOT be configuration problem.

You could review the wireshark log.

Thanks!

2012-12-03



sparklezou



????sparklezou
?????2012-12-02 17:03
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:

Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.


FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.

Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.

Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.

Then in the following whole process, there will be display name both for "from" & "to".

How about your idea?

Thanks in advance!

2012-12-02



sparklezou



????sparklezou
?????2012-11-30 21:13
???Re: Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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From enst.bupt at gmail.com  Tue Dec  4 15:12:02 2012
From: enst.bupt at gmail.com (Bing LI)
Date: Tue, 4 Dec 2012 07:12:02 -0500
Subject: [Freeswitch-dev] [Freeswitch-users] About the display problem
 during calling
In-Reply-To: <6e81da06.56bc.13b64a9644b.Coremail.sparklezou@163.com>
References: <65B9D5C2-3A8E-441C-ABFD-C6BB57FDA896@jerris.com>
	<b07ecce.5acd.13b616d67b3.Coremail.sparklezou@163.com>
	<6e81da06.56bc.13b64a9644b.Coremail.sparklezou@163.com>
Message-ID: <CAP9o=2mCsvE1xhk43sVadJiv+xysbCgZ1=TSM5b+BTYQ4be1zg@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_name
http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
http://wiki.freeswitch.org/wiki/Clarification:gateways#Caller_ID



On Tue, Dec 4, 2012 at 1:43 AM, sparklezou <sparklezou at 163.com> wrote:

> **
> Hi Mike & Brian,
>
> I  find the way to correct the final 200 SDP response message.
>
>  The "Outbound Call" is in the 200 SDP message. the called is set to
>
> ""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "
>
> In the switch_ivr_originate.c file,  in the following code.
>
>       new_profile->callee_id_name = switch_core_strdup(new_profile->pool,
> "Outbound Call");
>
> Before this sentence, there should be one function to find the display
> name from the number. Then replace the default "Outbound Call".
>
> Is there any function to find the name from the user xml files?
>
> In this way, just correct the final dispaly issue.
>
> And during the call, I still NOT find the corret location to add the
> dispaly name in to the sip "to" field.
>
> But I believe it should NOT add in the sofia sip stack side. It should add
> at the uplay application invote function side.
>
> I think, it should be in switch_channel.c switch_ivr_originate.c
> switch_event.c files.
>
> Please check and consider it.
>
> Thanks!
>
>
> 2012-12-04
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-12-03 23:40
> *???*Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display
> problem during calling
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*
>
>  Hi Mike,
>
> Thank!
>
> In which functin should be the right place to add the Name for the "to"
> field?
>
> How about your idea?
>
> 2012-12-03
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*Michael Jerris
> *?????*2012-12-03 23:20
> *???*Re: [Freeswitch-dev] [Freeswitch-users] About the display problem
> during calling
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*
>
> All of this is handled in mod_sofia, look for a function with i_invite in
> the name.
>
> Mike
>
>  On Dec 3, 2012, at 4:30 AM, sparklezou <sparklezou at 163.com> wrote:
>
>   Hi Brian,
>
> I'm trying to find where the session start.
>
> Just found it start from switch_channel.c:951, the new channel. There is
> NO more debug info.
>
> My idea, when get the INVITE SIP message from the sofia stack. check the
> called number detail info in FS. add the dispaly name in the "to" field of
> the following SIP message.
>
> It should finish before the new channel.
>
> where could add such code?
>
> Thanks!
>
>
> 2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel
> sofia/internal/1001 at 172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc]
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (
> sofia/internal/1001 at 172.0.0.10) Running State Change CS_NEW
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (
> sofia/internal/1001 at 172.0.0.10) State NEW
> 2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by
> acl "domains". Falling back to Digest auth.
> 2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session
> b51ac8ed-69d4-409a-95ba-58b420c77bfc
> 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session
> b51ac8ed-69d4-409a-95ba-58b420c77bfc
> 2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal
> sofia/internal/1001 at 172.0.0.10 [BREAK]
> 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by
> acl "domains". Falling back to Digest auth.
> 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel
> sofia/internal/1001 at 172.0.0.10 entering state [received][100]
> 2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:
>
> 2012-12-03
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-12-03 09:53
> *???*Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display
> problem during calling
> *????*"brian"<brian at freeswitch.org>
> *???*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
>
>  Hi Brian,
>
> It's the default Dialplan.
>
> I have tried the fresh last version also. The same result. It should NOT
> be configuration problem.
>
> You could review the wireshark log.
>
> Thanks!
>
> 2012-12-03
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-12-02 17:03
> *???*Re: [Freeswitch-dev] [Freeswitch-users] About the display problem
> during calling
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*
>
>  Hi Develop team,
>
> In these days, I reviewed the "DEBUG" info, and the sip info log.
> summarize as following:
>
> Caller  <---> FS, in the "from" with dispaly neam. The display is set from
> the Caller Phone. in the "to" without the display name. As the caller only
> dial the number.
>
>
> FS -----> Called, in the "from" with display name.It's transfer from
> "Caller  <---> FS". in the "to" without the display neme. As the caller
> only dial the number.
>
> Called  -----> FS. in the "to" with display name. Here the Called Phone
> add the Display name into the SIP message.
>
> Here conside the "Caller <----> FS"  and "FS<---->Called" are two
> sessions. I think the best way, is check the Display Name of Called in FS
> system when get the INVITE message. Add the Display Name for the called in
> the whole process.
>
> Then in the following whole process, there will be display name both for
> "from" & "to".
>
> How about your idea?
>
> Thanks in advance!
>
> 2012-12-02
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-11-30 21:13
> *???*Re: Re: [Freeswitch-users] About the display problem during calling
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*
>
>  Hi Develop team,
>
> Just simlify this question.
>
> In the message 100, 180,183, 200 response to the caller side from
> Freeswitch side, the called part Name should be include in the sip address
> of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only
> <sip: 1234 at a.b.c.d>.
>
> Currently, at the called part, the caller info is correct with Name. So
> the display at called side is OK.
>
> Could you please let me know where is the source code for such process? I
> also would like to review it.
>
> Thanks!
>
> 2012-11-30
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-11-30 15:55
> *???*Re: [Freeswitch-users] About the display problem during calling
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*"freeswitch-users"<freeswitch-users at lists.freeswitch.org>
>
>  Hi All,
>
> Before, I saved the user name in the phone. So it will show the name when
> ringing.
>
> From the FreeSwitch side, there is NO difference.
>
> @ Develop Team,
>
> For Internal call, or any sip call from gateway, it should be better show
> the Name during ring.  And also keep show the name in the call, in stead of
> "Outbound Call".
>
> Usually the phone will update the display name from the sip response
> message.
>
> So in the 100, 180, 183, 200 sip/sdp message. Respons with the called
> name, like "ABC"<sip: 1234 at a.b.c.d>.
>
> Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In
> the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>;
> ....... "
>
> If it could be updated, then the called name could be displayed on the
> caller phone.
>
> Thanks!
>
> 2012-11-30
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-11-30 11:19
> *???*[Freeswitch-users] About the display problem during calling
> *????*"freeswitch-users"<freeswitch-users at lists.freeswitch.org
> >,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*
>
> **
> Hi All,
>
> I facing a strange problem.
>
> Before, when calling Internal number, the name and number will both be
> displayed on the caller phone. When connected, only the name will display
> on the phone. At called phone, display the name and number. Seems fine.
>
> But yesterday afternoon, when calling Internal number, only the number is
> displayed on the caller phone. When connected, display "Outbound Call" on
> the caller phone. At the called phone, everthing is the same. Display both
> the name and number.
>
> I think, mabybe something configured wrong. So I try to restore back the
> configuration. seems the same. I have to re-install the last release stable
> 1.2.5.1 version on another server. total fresh setting. But found the same
> problem.
>
> I checked the sip info.
>
> The "Outbound Call" is in the 200 SDP message. the called is set to
>
> ""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "
>
> Not the correct called name.
>
> Is there anything wrong?
>
> Please help me.
>
> Thanks!
>
>
> 2012-11-30
> ------------------------------
>  sparklezou
> **
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From sparklezou at 163.com  Wed Dec  5 11:35:48 2012
From: sparklezou at 163.com (sparklezou)
Date: Wed, 5 Dec 2012 16:35:48 +0800
Subject: [Freeswitch-dev] [Freeswitch-users] About the display problem
 during calling
In-Reply-To: <CAP9o=2mCsvE1xhk43sVadJiv+xysbCgZ1=TSM5b+BTYQ4be1zg@mail.gmail.com>
References: <CAP9o=2mCsvE1xhk43sVadJiv+xysbCgZ1=TSM5b+BTYQ4be1zg@mail.gmail.com>
Message-ID: <1cf3b8cf.3911.13b6a370057.Coremail.sparklezou@163.com>

Hi Li Bing,

Thanks! I need is effective_callee_id_number

@Develop team,

After set the following parameters in the dialplan

<action application="set" data="effective_callee_id_name= ${user_data(${dialed_extension}@${domain_name} var effective_caller_id_number)}" />

The Callee Name is showed in the field of "Remote-Part-ID: " in 183 SIP message.

But it still NOT appear in the "to" field in SIP message. Aslo NOT in the 200 "Remote-Part-ID: ".

For "200" SIP message, it should be update the following line code in switch_ivr_originate.c.
 
      new_profile->callee_id_name = switch_core_strdup(new_profile->pool, "Outbound Call");

It should be something like 

  new_profile->callee_id_number = switch_sanitize_number(switch_core_strdup(new_profile->pool, new_profile->destination_number));

But I could NOT find the correct parameter.

For the "to" field, where should be the correct location to update it?

Thanks!

2012-12-05



sparklezou



????Bing LI
?????2012-12-04 20:12
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_name
http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
http://wiki.freeswitch.org/wiki/Clarification:gateways#Caller_ID





On Tue, Dec 4, 2012 at 1:43 AM, sparklezou <sparklezou at 163.com> wrote:

Hi Mike & Brian,

I  find the way to correct the final 200 SDP response message.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

In the switch_ivr_originate.c file,  in the following code.
    
      new_profile->callee_id_name = switch_core_strdup(new_profile->pool, "Outbound Call");

Before this sentence, there should be one function to find the display name from the number. Then replace the default "Outbound Call".

Is there any function to find the name from the user xml files?

In this way, just correct the final dispaly issue.

And during the call, I still NOT find the corret location to add the dispaly name in to the sip "to" field.

But I believe it should NOT add in the sofia sip stack side. It should add at the uplay application invote function side.

I think, it should be in switch_channel.c switch_ivr_originate.c switch_event.c files.

Please check and consider it.

Thanks!


2012-12-04



sparklezou



????sparklezou
?????2012-12-03 23:40
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Mike,

Thank!

In which functin should be the right place to add the Name for the "to" field? 

How about your idea?

2012-12-03



sparklezou



????Michael Jerris
?????2012-12-03 23:20
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

All of this is handled in mod_sofia, look for a function with i_invite in the name. 


Mike


On Dec 3, 2012, at 4:30 AM, sparklezou <sparklezou at 163.com> wrote:


Hi Brian,

I'm trying to find where the session start.

Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.

My idea, when get the INVITE SIP message from the sofia stack. check the called number detail info in FS. add the dispaly name in the "to" field of the following SIP message.

It should finish before the new channel.

where could add such code?

Thanks!


2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel sofia/internal/1001 at 172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1001 at 172.0.0.10) Running State Change CS_NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1001 at 172.0.0.10) State NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel sofia/internal/1001 at 172.0.0.10 entering state [received][100]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:

2012-12-03



sparklezou



????sparklezou
?????2012-12-03 09:53
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"brian"<brian at freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Brian,

It's the default Dialplan.

I have tried the fresh last version also. The same result. It should NOT be configuration problem.

You could review the wireshark log.

Thanks!

2012-12-03



sparklezou



????sparklezou
?????2012-12-02 17:03
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:

Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.


FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.

Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.

Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.

Then in the following whole process, there will be display name both for "from" & "to".

How about your idea?

Thanks in advance!

2012-12-02



sparklezou



????sparklezou
?????2012-11-30 21:13
???Re: Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org
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From sparklezou at 163.com  Wed Dec  5 11:39:05 2012
From: sparklezou at 163.com (sparklezou)
Date: Wed, 5 Dec 2012 16:39:05 +0800
Subject: [Freeswitch-dev] How to show Callee's Name on Caller's phone during
	call
In-Reply-To: <1cf3b8cf.3911.13b6a370057.Coremail.sparklezou@163.com>
References: <CAP9o=2mCsvE1xhk43sVadJiv+xysbCgZ1=TSM5b+BTYQ4be1zg@mail.gmail.com><1cf3b8cf.3911.13b6a370057.Coremail.sparklezou@163.com>
Message-ID: <6b9318d5.3977.13b6a3a0214.Coremail.sparklezou@163.com>

Hi Develop team,

Just update the title. It will be much better.

After set the following parameters in the dialplan

<action application="set" data="effective_callee_id_name= ${user_data(${dialed_extension}@${domain_name} var effective_caller_id_number)}" />

The Callee Name is showed in the field of "Remote-Part-ID: " in 183 SIP message.

But it still NOT appear in the "to" field in SIP message. Aslo NOT in the 200 "Remote-Part-ID: ".

For "200" SIP message, it should be update the following line code in switch_ivr_originate.c.

      new_profile->callee_id_name = switch_core_strdup(new_profile->pool, "Outbound Call");

It should be something like 

  new_profile->callee_id_number = switch_sanitize_number(switch_core_strdup(new_profile->pool, new_profile->destination_number));

But I could NOT find the correct parameter.

For the "to" field, where should be the correct location to update it?

Thanks!

2012-12-05



sparklezou



????Bing LI
?????2012-12-04 20:12
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_name 
http://wiki.freeswitch.org/wiki/Variable_effective_caller_id_number
http://wiki.freeswitch.org/wiki/Clarification:gateways#Caller_ID





On Tue, Dec 4, 2012 at 1:43 AM, sparklezou <sparklezou at 163.com> wrote:

Hi Mike & Brian,

I  find the way to correct the final 200 SDP response message.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

In the switch_ivr_originate.c file,  in the following code.
    
      new_profile->callee_id_name = switch_core_strdup(new_profile->pool, "Outbound Call");

Before this sentence, there should be one function to find the display name from the number. Then replace the default "Outbound Call".

Is there any function to find the name from the user xml files?

In this way, just correct the final dispaly issue.

And during the call, I still NOT find the corret location to add the dispaly name in to the sip "to" field.

But I believe it should NOT add in the sofia sip stack side. It should add at the uplay application invote function side.

I think, it should be in switch_channel.c switch_ivr_originate.c switch_event.c files.

Please check and consider it.

Thanks!


2012-12-04



sparklezou



????sparklezou
?????2012-12-03 23:40
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Mike,

Thank!

In which functin should be the right place to add the Name for the "to" field? 

How about your idea?

2012-12-03



sparklezou



????Michael Jerris
?????2012-12-03 23:20
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

All of this is handled in mod_sofia, look for a function with i_invite in the name. 


Mike


On Dec 3, 2012, at 4:30 AM, sparklezou <sparklezou at 163.com> wrote:


Hi Brian,

I'm trying to find where the session start.

Just found it start from switch_channel.c:951, the new channel. There is NO more debug info.

My idea, when get the INVITE SIP message from the sofia stack. check the called number detail info in FS. add the dispaly name in the "to" field of the following SIP message.

It should finish before the new channel.

where could add such code?

Thanks!


2012-12-02 16:15:36.917452 [NOTICE] switch_channel.c:951 New Channel sofia/internal/1001 at 172.0.0.10 [b51ac8ed-69d4-409a-95ba-58b420c77bfc]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:415 (sofia/internal/1001 at 172.0.0.10) Running State Change CS_NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/1001 at 172.0.0.10) State NEW
2012-12-02 16:15:36.917452 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:36.937457 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:36.937457 [DEBUG] sofia.c:1733 detaching session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:1825 Re-attaching to session b51ac8ed-69d4-409a-95ba-58b420c77bfc
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1001 at 172.0.0.10 [BREAK]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:8435 IP 172.0.0.129 Rejected by acl "domains". Falling back to Digest auth.
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6306 Channel sofia/internal/1001 at 172.0.0.10 entering state [received][100]
2012-12-02 16:15:37.077884 [DEBUG] sofia.c:6317 Remote SDP:

2012-12-03



sparklezou



????sparklezou
?????2012-12-03 09:53
???Re: Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"brian"<brian at freeswitch.org>
???"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>

Hi Brian,

It's the default Dialplan.

I have tried the fresh last version also. The same result. It should NOT be configuration problem.

You could review the wireshark log.

Thanks!

2012-12-03



sparklezou



????sparklezou
?????2012-12-02 17:03
???Re: [Freeswitch-dev] [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

In these days, I reviewed the "DEBUG" info, and the sip info log. summarize as following:

Caller  <---> FS, in the "from" with dispaly neam. The display is set from the Caller Phone. in the "to" without the display name. As the caller only dial the number.


FS -----> Called, in the "from" with display name.It's transfer from "Caller  <---> FS". in the "to" without the display neme. As the caller only dial the number.

Called  -----> FS. in the "to" with display name. Here the Called Phone add the Display name into the SIP message.

Here conside the "Caller <----> FS"  and "FS<---->Called" are two sessions. I think the best way, is check the Display Name of Called in FS system when get the INVITE message. Add the Display Name for the called in the whole process.

Then in the following whole process, there will be display name both for "from" & "to".

How about your idea?

Thanks in advance!

2012-12-02



sparklezou



????sparklezou
?????2012-11-30 21:13
???Re: Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi Develop team,

Just simlify this question.

In the message 100, 180,183, 200 response to the caller side from Freeswitch side, the called part Name should be include in the sip address of "To" & SDP part. It should be  "ABC"<sip: 1234 at a.b.c.d>, NOT only <sip: 1234 at a.b.c.d>.

Currently, at the called part, the caller info is correct with Name. So the display at called side is OK.

Could you please let me know where is the source code for such process? I also would like to review it.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 15:55
???Re: [Freeswitch-users] About the display problem during calling
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???"freeswitch-users"<freeswitch-users at lists.freeswitch.org>

Hi All,

Before, I saved the user name in the phone. So it will show the name when ringing.

From the FreeSwitch side, there is NO difference.

@ Develop Team,

For Internal call, or any sip call from gateway, it should be better show the Name during ring.  And also keep show the name in the call, in stead of "Outbound Call".

Usually the phone will update the display name from the sip response message.

So in the 100, 180, 183, 200 sip/sdp message. Respons with the called name, like "ABC"<sip: 1234 at a.b.c.d>.

Currently in the inital 100,180,183 message, only <sip: 1234 at a.b.c.d>. In the 200 message "Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

If it could be updated, then the called name could be displayed on the caller phone.

Thanks!

2012-11-30



sparklezou



????sparklezou
?????2012-11-30 11:19
???[Freeswitch-users] About the display problem during calling
????"freeswitch-users"<freeswitch-users at lists.freeswitch.org>,"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

I facing a strange problem.

Before, when calling Internal number, the name and number will both be displayed on the caller phone. When connected, only the name will display on the phone. At called phone, display the name and number. Seems fine.

But yesterday afternoon, when calling Internal number, only the number is displayed on the caller phone. When connected, display "Outbound Call" on the caller phone. At the called phone, everthing is the same. Display both the name and number.

I think, mabybe something configured wrong. So I try to restore back the configuration. seems the same. I have to re-install the last release stable 1.2.5.1 version on another server. total fresh setting. But found the same problem.

I checked the sip info.

The "Outbound Call" is in the 200 SDP message. the called is set to

""Remote-Part-ID: "Outbound Call" <sip: 1234 at a.b.c.d>; ....... "

Not the correct called name.

Is there anything wrong?

Please help me.

Thanks!


2012-11-30



sparklezou
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org
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From msc at freeswitch.org  Wed Dec  5 20:05:12 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 09:05:12 -0800
Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxUeDLWVF23J_9RZR+KfE0EQmY4koDw2DpNhWKz9F1G0-g@mail.gmail.com>

Hello folks,

We'll be having our conference call in just about one hour. The agenda page
is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05

We'll be discussing some interesting facts about the FreeSWITCH XML
pre-processor and also an object lesson in how to use the source code to
answer a question and then put it on the wiki. Oh, and we have an update on
the ClueCon videos!

Talk to you soon.

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec  5 21:06:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Dec 2012 10:06:59 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Subject: Windows
	debugging tools
Message-ID: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>

Hey all!

I'm trying to gauge the interest level in this subject. We have an
experienced Windows user who is willing to share with us a lot of his
hard-earned knowledge when it comes to debugging crashes and such in a
Windows environment. If you are interested in hearing about this subject
please respond. (Only respond if you are interested - we don't need to hear
from those who are not interested.)

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From v.kovalyshyn at gmail.com  Wed Dec  5 21:44:55 2012
From: v.kovalyshyn at gmail.com (Vitaly Kovalyshyn)
Date: Wed, 5 Dec 2012 20:44:55 +0200
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Subject: Windows
	debugging tools
In-Reply-To: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>
References: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>
Message-ID: <B92DCCA9-C569-4C76-B71C-4A107B557E6C@gmail.com>

Hi!

I'm interested! Because, we made some soft based on FS, that's work on Windows.

__
Vitaly Kovalyshyn
Twitter: @kovalyshyn
http://vk.it-sfera.com.ua/ 
http://wiki.webitel.com/

????????? ? iPad

5 ????. 2012 ? 20:06 Michael Collins <msc at freeswitch.org> ???????(??):

> Hey all!
> 
> I'm trying to gauge the interest level in this subject. We have an experienced Windows user who is willing to share with us a lot of his hard-earned knowledge when it comes to debugging crashes and such in a Windows environment. If you are interested in hearing about this subject please respond. (Only respond if you are interested - we don't need to hear from those who are not interested.)
> 
> Thanks!
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
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From vutamhoan at gmail.com  Fri Dec  7 04:45:07 2012
From: vutamhoan at gmail.com (Vu Quang Hoa)
Date: Fri, 7 Dec 2012 08:45:07 +0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Subject: Windows
 debugging tools
In-Reply-To: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>
References: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>
Message-ID: <CAJnMD2c==qAegsSmH+8poXrGx+_BqD--WNJ8nJfNtJ+iV3mquw@mail.gmail.com>

Add me to the list, please!

Thank you!
vutamhoan

On Thu, Dec 6, 2012 at 1:06 AM, Michael Collins <msc at freeswitch.org> wrote:

> Hey all!
>
> I'm trying to gauge the interest level in this subject. We have an
> experienced Windows user who is willing to share with us a lot of his
> hard-earned knowledge when it comes to debugging crashes and such in a
> Windows environment. If you are interested in hearing about this subject
> please respond. (Only respond if you are interested - we don't need to hear
> from those who are not interested.)
>
> Thanks!
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From eduardonunesp at gmail.com  Mon Dec 10 18:57:57 2012
From: eduardonunesp at gmail.com (Eduardo Nunes Pereira)
Date: Mon, 10 Dec 2012 13:57:57 -0200
Subject: [Freeswitch-dev] About call_cause with switch_channel_hangup
Message-ID: <CAAXv2ZL9hDrKYkDJ3ncdr8iQGdP46PK7DjSp1cuRhdU7VHLrTw@mail.gmail.com>

I have some problem with my module , when channel hangup
(switch_channel_hangup) with the cause specified like cause
DESTINATION_OUT_OF_ORDER, the other channel in bridge receives a
NORMAL_CLEARING, but when the hangup was made in a channel that is not
completed bridged (not answered yet) the cause was
passed DESTINATION_OUT_OF_ORDER (cause that i want)



-- 
Eduardo Nunes Pereira
skype: eduardonunesp
msn:eduardonunesp
http://about.me/eduardonunesp
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From niels.thomsen at fentechnology.co.uk  Mon Dec 10 19:17:09 2012
From: niels.thomsen at fentechnology.co.uk (Niels Thomsen)
Date: Mon, 10 Dec 2012 16:17:09 +0000
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Subject:
	Windowsdebugging tools
In-Reply-To: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>
References: <CAKzWOxVKEBkaACC2q=SV9izD=ymaoUcTQxM0CyvCKiWTNUxSLg@mail.gmail.com>
Message-ID: <A4F410A7C1D87646A6E2039559A50D7A62FA74@GIMLI.fen.local>

Yes please!

 

- Niels

 

From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 05 December 2012 18:07
To: freeswitch-users at lists.freeswitch.org;
freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Subject:
Windowsdebugging tools

 

Hey all!

I'm trying to gauge the interest level in this subject. We have an
experienced Windows user who is willing to share with us a lot of his
hard-earned knowledge when it comes to debugging crashes and such in a
Windows environment. If you are interested in hearing about this subject
please respond. (Only respond if you are interested - we don't need to
hear from those who are not interested.)

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org



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From marketing at cluecon.com  Mon Dec 10 23:29:01 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 10 Dec 2012 12:29:01 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>

Greetings!

We are glad to report that the FreeSWITCH team has tagged version 1.2.5.3.
You can download the tarball
here<http://files.freeswitch.org/freeswitch-1.2.5.3.tar.bz2>.
Anyone using 1.2.5.x should update as soon as possible. We appreciate all
those who have helped us with testing and tracking down some sneaky and
pernicious little bugs.

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_05>we spent
some time talking about the XML parser and some of its
pre-processor directives. We discussed specifically how you can use the
"exec" command to execute a shell script in the middle of XML processing.
We also discussed a few tricks on how to look at the source code when you
need to learn about some FreeSWITCH functionality that otherwise is not
documented. This week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12>subject is
still pending, so stay tuned!

One other item I'd like to mention is that we've had several reports of
FreeSWITCH success stories. We will be providing more information about
those in upcoming stories on our Web site. We've got people using
FreeSWITCH in various situations as well as software developers who've
added support for FreeSWITCH to their offerings. The FreeSWITCH ecosystem
continues to grow and flourish! Thank you all for being a part of it.

Take care and have a great week!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From kris at kriskinc.com  Mon Dec 10 23:55:07 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 10 Dec 2012 15:55:07 -0500
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
References: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
Message-ID: <CAKDfjge4oCMwVR17n4qSpEWSAi9tsUhi+zb0T4YAmuNCrwBRPg@mail.gmail.com>

FYI 1.2.5.2 and 1.2.5.3 still aren't available on files.freeswitch.org :(

On Mon, Dec 10, 2012 at 3:29 PM, Michael Collins <marketing at cluecon.com> wrote:
> Greetings!
>
> We are glad to report that the FreeSWITCH team has tagged version 1.2.5.3.
> You can download the tarball here. Anyone using 1.2.5.x should update as
> soon as possible. We appreciate all those who have helped us with testing
> and tracking down some sneaky and pernicious little bugs.
>
> On last week's conference call we spent some time talking about the XML
> parser and some of its pre-processor directives. We discussed specifically
> how you can use the "exec" command to execute a shell script in the middle
> of XML processing. We also discussed a few tricks on how to look at the
> source code when you need to learn about some FreeSWITCH functionality that
> otherwise is not documented. This week's conference call subject is still
> pending, so stay tuned!
>
> One other item I'd like to mention is that we've had several reports of
> FreeSWITCH success stories. We will be providing more information about
> those in upcoming stories on our Web site. We've got people using FreeSWITCH
> in various situations as well as software developers who've added support
> for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues to
> grow and flourish! Thank you all for being a part of it.
>
> Take care and have a great week!
>
> --
> Michael S Collins
> ClueCon Team
> http://www.cluecon.com
> 877-7-4ACLUE
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From msc at freeswitch.org  Tue Dec 11 00:12:22 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 13:12:22 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKDfjge4oCMwVR17n4qSpEWSAi9tsUhi+zb0T4YAmuNCrwBRPg@mail.gmail.com>
References: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
	<CAKDfjge4oCMwVR17n4qSpEWSAi9tsUhi+zb0T4YAmuNCrwBRPg@mail.gmail.com>
Message-ID: <CAKzWOxUQet=H7fCsyh9gygbELWxmF6hcpsYvCgFcs6DwkWxYQA@mail.gmail.com>

Actually, Mr. Rice has said that he would get them up on the list. However,
I think if you actually click the link it will grab the file. At least it
did for me about 10 seconds ago...

-MC

On Mon, Dec 10, 2012 at 12:55 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> FYI 1.2.5.2 and 1.2.5.3 still aren't available on files.freeswitch.org :(
>
> On Mon, Dec 10, 2012 at 3:29 PM, Michael Collins <marketing at cluecon.com>
> wrote:
> > Greetings!
> >
> > We are glad to report that the FreeSWITCH team has tagged version
> 1.2.5.3.
> > You can download the tarball here. Anyone using 1.2.5.x should update as
> > soon as possible. We appreciate all those who have helped us with testing
> > and tracking down some sneaky and pernicious little bugs.
> >
> > On last week's conference call we spent some time talking about the XML
> > parser and some of its pre-processor directives. We discussed
> specifically
> > how you can use the "exec" command to execute a shell script in the
> middle
> > of XML processing. We also discussed a few tricks on how to look at the
> > source code when you need to learn about some FreeSWITCH functionality
> that
> > otherwise is not documented. This week's conference call subject is still
> > pending, so stay tuned!
> >
> > One other item I'd like to mention is that we've had several reports of
> > FreeSWITCH success stories. We will be providing more information about
> > those in upcoming stories on our Web site. We've got people using
> FreeSWITCH
> > in various situations as well as software developers who've added support
> > for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues to
> > grow and flourish! Thank you all for being a part of it.
> >
> > Take care and have a great week!
> >
> > --
> > Michael S Collins
> > ClueCon Team
> > http://www.cluecon.com
> > 877-7-4ACLUE
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
>
>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From kris at kriskinc.com  Tue Dec 11 01:08:41 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 10 Dec 2012 17:08:41 -0500
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKzWOxUQet=H7fCsyh9gygbELWxmF6hcpsYvCgFcs6DwkWxYQA@mail.gmail.com>
References: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
	<CAKDfjge4oCMwVR17n4qSpEWSAi9tsUhi+zb0T4YAmuNCrwBRPg@mail.gmail.com>
	<CAKzWOxUQet=H7fCsyh9gygbELWxmF6hcpsYvCgFcs6DwkWxYQA@mail.gmail.com>
Message-ID: <CAKDfjgdXwSTg-xNK-gKOjdBfehFBh6TP0Aep7U2Jdh722=+_5A@mail.gmail.com>

404 for me

On Mon, Dec 10, 2012 at 4:12 PM, Michael Collins <msc at freeswitch.org> wrote:
> Actually, Mr. Rice has said that he would get them up on the list. However,
> I think if you actually click the link it will grab the file. At least it
> did for me about 10 seconds ago...
>
> -MC
>
>
> On Mon, Dec 10, 2012 at 12:55 PM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
>>
>> FYI 1.2.5.2 and 1.2.5.3 still aren't available on files.freeswitch.org :(
>>
>> On Mon, Dec 10, 2012 at 3:29 PM, Michael Collins <marketing at cluecon.com>
>> wrote:
>> > Greetings!
>> >
>> > We are glad to report that the FreeSWITCH team has tagged version
>> > 1.2.5.3.
>> > You can download the tarball here. Anyone using 1.2.5.x should update as
>> > soon as possible. We appreciate all those who have helped us with
>> > testing
>> > and tracking down some sneaky and pernicious little bugs.
>> >
>> > On last week's conference call we spent some time talking about the XML
>> > parser and some of its pre-processor directives. We discussed
>> > specifically
>> > how you can use the "exec" command to execute a shell script in the
>> > middle
>> > of XML processing. We also discussed a few tricks on how to look at the
>> > source code when you need to learn about some FreeSWITCH functionality
>> > that
>> > otherwise is not documented. This week's conference call subject is
>> > still
>> > pending, so stay tuned!
>> >
>> > One other item I'd like to mention is that we've had several reports of
>> > FreeSWITCH success stories. We will be providing more information about
>> > those in upcoming stories on our Web site. We've got people using
>> > FreeSWITCH
>> > in various situations as well as software developers who've added
>> > support
>> > for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues to
>> > grow and flourish! Thank you all for being a part of it.
>> >
>> > Take care and have a great week!
>> >
>> > --
>> > Michael S Collins
>> > ClueCon Team
>> > http://www.cluecon.com
>> > 877-7-4ACLUE
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From msc at freeswitch.org  Tue Dec 11 01:42:16 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 14:42:16 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKDfjgdXwSTg-xNK-gKOjdBfehFBh6TP0Aep7U2Jdh722=+_5A@mail.gmail.com>
References: <CAKzWOxWG_5d5_jBVB-2H-VShLOmU8UB=BWqkhW4h4yywsAfKZg@mail.gmail.com>
	<CAKDfjge4oCMwVR17n4qSpEWSAi9tsUhi+zb0T4YAmuNCrwBRPg@mail.gmail.com>
	<CAKzWOxUQet=H7fCsyh9gygbELWxmF6hcpsYvCgFcs6DwkWxYQA@mail.gmail.com>
	<CAKDfjgdXwSTg-xNK-gKOjdBfehFBh6TP0Aep7U2Jdh722=+_5A@mail.gmail.com>
Message-ID: <CAKzWOxWV7p9VJOAejBNq7M3L_ZDJo4Uzj77FP5=+46XA8FOynw@mail.gmail.com>

Interesting. Ken is already checking on that. It's probably a CDN issue.
We'll ping everyone when the files site is updated.
-MC

On Mon, Dec 10, 2012 at 2:08 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> 404 for me
>
> On Mon, Dec 10, 2012 at 4:12 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> > Actually, Mr. Rice has said that he would get them up on the list.
> However,
> > I think if you actually click the link it will grab the file. At least it
> > did for me about 10 seconds ago...
> >
> > -MC
> >
> >
> > On Mon, Dec 10, 2012 at 12:55 PM, Kristian Kielhofner <kris at kriskinc.com
> >
> > wrote:
> >>
> >> FYI 1.2.5.2 and 1.2.5.3 still aren't available on files.freeswitch.org:(
> >>
> >> On Mon, Dec 10, 2012 at 3:29 PM, Michael Collins <marketing at cluecon.com
> >
> >> wrote:
> >> > Greetings!
> >> >
> >> > We are glad to report that the FreeSWITCH team has tagged version
> >> > 1.2.5.3.
> >> > You can download the tarball here. Anyone using 1.2.5.x should update
> as
> >> > soon as possible. We appreciate all those who have helped us with
> >> > testing
> >> > and tracking down some sneaky and pernicious little bugs.
> >> >
> >> > On last week's conference call we spent some time talking about the
> XML
> >> > parser and some of its pre-processor directives. We discussed
> >> > specifically
> >> > how you can use the "exec" command to execute a shell script in the
> >> > middle
> >> > of XML processing. We also discussed a few tricks on how to look at
> the
> >> > source code when you need to learn about some FreeSWITCH functionality
> >> > that
> >> > otherwise is not documented. This week's conference call subject is
> >> > still
> >> > pending, so stay tuned!
> >> >
> >> > One other item I'd like to mention is that we've had several reports
> of
> >> > FreeSWITCH success stories. We will be providing more information
> about
> >> > those in upcoming stories on our Web site. We've got people using
> >> > FreeSWITCH
> >> > in various situations as well as software developers who've added
> >> > support
> >> > for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues
> to
> >> > grow and flourish! Thank you all for being a part of it.
> >> >
> >> > Take care and have a great week!
> >> >
> >> > --
> >> > Michael S Collins
> >> > ClueCon Team
> >> > http://www.cluecon.com
> >> > 877-7-4ACLUE
> >> >
> >> >
> >> >
> >> >
> _________________________________________________________________________
> >> > Professional FreeSWITCH Consulting Services:
> >> > consulting at freeswitch.org
> >> > http://www.freeswitchsolutions.com
> >> >
> >> > 
> >> > 
> >> >
> >> > Official FreeSWITCH Sites
> >> > http://www.freeswitch.org
> >> > http://wiki.freeswitch.org
> >> > http://www.cluecon.com
> >> >
> >> > FreeSWITCH-dev mailing list
> >> > FreeSWITCH-dev at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> > http://www.freeswitch.org
> >> >
> >>
> >>
> >>
> >> --
> >> Kristian Kielhofner
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> FreeSWITCH-dev mailing list
> >> FreeSWITCH-dev at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> >> http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Michael S Collins
> > Twitter: @mercutioviz
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
>
>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From krice at freeswitch.org  Tue Dec 11 01:44:20 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 10 Dec 2012 16:44:20 -0600
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKDfjgdXwSTg-xNK-gKOjdBfehFBh6TP0Aep7U2Jdh722=+_5A@mail.gmail.com>
Message-ID: <CCEBC1E4.61D62%krice@freeswitch.org>

http://files.freeswitch.org/freeswitch-1.2.5.3.tar.bz2  not working?

Ok the CDN is starting to get on my nerves

K


On 12/10/12 4:08 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:

> 404 for me
> 
> On Mon, Dec 10, 2012 at 4:12 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Actually, Mr. Rice has said that he would get them up on the list. However,
>> I think if you actually click the link it will grab the file. At least it
>> did for me about 10 seconds ago...
>> 
>> -MC
>> 
>> 
>> On Mon, Dec 10, 2012 at 12:55 PM, Kristian Kielhofner <kris at kriskinc.com>
>> wrote:
>>> 
>>> FYI 1.2.5.2 and 1.2.5.3 still aren't available on files.freeswitch.org :(
>>> 
>>> On Mon, Dec 10, 2012 at 3:29 PM, Michael Collins <marketing at cluecon.com>
>>> wrote:
>>>> Greetings!
>>>> 
>>>> We are glad to report that the FreeSWITCH team has tagged version
>>>> 1.2.5.3.
>>>> You can download the tarball here. Anyone using 1.2.5.x should update as
>>>> soon as possible. We appreciate all those who have helped us with
>>>> testing
>>>> and tracking down some sneaky and pernicious little bugs.
>>>> 
>>>> On last week's conference call we spent some time talking about the XML
>>>> parser and some of its pre-processor directives. We discussed
>>>> specifically
>>>> how you can use the "exec" command to execute a shell script in the
>>>> middle
>>>> of XML processing. We also discussed a few tricks on how to look at the
>>>> source code when you need to learn about some FreeSWITCH functionality
>>>> that
>>>> otherwise is not documented. This week's conference call subject is
>>>> still
>>>> pending, so stay tuned!
>>>> 
>>>> One other item I'd like to mention is that we've had several reports of
>>>> FreeSWITCH success stories. We will be providing more information about
>>>> those in upcoming stories on our Web site. We've got people using
>>>> FreeSWITCH
>>>> in various situations as well as software developers who've added
>>>> support
>>>> for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues to
>>>> grow and flourish! Thank you all for being a part of it.
>>>> 
>>>> Take care and have a great week!
>>>> 
>>>> --
>>>> Michael S Collins
>>>> ClueCon Team
>>>> http://www.cluecon.com
>>>> 877-7-4ACLUE
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>>> 
>>> --
>>> Kristian Kielhofner
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>> 
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> --
>> Michael S Collins
>> Twitter: @mercutioviz
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>> 
> 
> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From kris at kriskinc.com  Tue Dec 11 01:52:03 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 10 Dec 2012 17:52:03 -0500
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
In-Reply-To: <CCEBC1E4.61D62%krice@freeswitch.org>
References: <CAKDfjgdXwSTg-xNK-gKOjdBfehFBh6TP0Aep7U2Jdh722=+_5A@mail.gmail.com>
	<CCEBC1E4.61D62%krice@freeswitch.org>
Message-ID: <CAKDfjgfyLaAJm80dZugiYNBqBzdTBG_jsh=gT17TfCcVXWnkaA@mail.gmail.com>

1.2.5.3 works for me, 1.2.5.2 does not (but I guess it doesn't matter
much anymore).

On Mon, Dec 10, 2012 at 5:44 PM, Ken Rice <krice at freeswitch.org> wrote:
> http://files.freeswitch.org/freeswitch-1.2.5.3.tar.bz2  not working?
>
> Ok the CDN is starting to get on my nerves
>
> K
>
>
> On 12/10/12 4:08 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:
>
>> 404 for me
>>
>> On Mon, Dec 10, 2012 at 4:12 PM, Michael Collins <msc at freeswitch.org> wrote:
>>> Actually, Mr. Rice has said that he would get them up on the list. However,
>>> I think if you actually click the link it will grab the file. At least it
>>> did for me about 10 seconds ago...
>>>
>>> -MC
>>>
>>>
>>> On Mon, Dec 10, 2012 at 12:55 PM, Kristian Kielhofner <kris at kriskinc.com>
>>> wrote:
>>>>
>>>> FYI 1.2.5.2 and 1.2.5.3 still aren't available on files.freeswitch.org :(
>>>>
>>>> On Mon, Dec 10, 2012 at 3:29 PM, Michael Collins <marketing at cluecon.com>
>>>> wrote:
>>>>> Greetings!
>>>>>
>>>>> We are glad to report that the FreeSWITCH team has tagged version
>>>>> 1.2.5.3.
>>>>> You can download the tarball here. Anyone using 1.2.5.x should update as
>>>>> soon as possible. We appreciate all those who have helped us with
>>>>> testing
>>>>> and tracking down some sneaky and pernicious little bugs.
>>>>>
>>>>> On last week's conference call we spent some time talking about the XML
>>>>> parser and some of its pre-processor directives. We discussed
>>>>> specifically
>>>>> how you can use the "exec" command to execute a shell script in the
>>>>> middle
>>>>> of XML processing. We also discussed a few tricks on how to look at the
>>>>> source code when you need to learn about some FreeSWITCH functionality
>>>>> that
>>>>> otherwise is not documented. This week's conference call subject is
>>>>> still
>>>>> pending, so stay tuned!
>>>>>
>>>>> One other item I'd like to mention is that we've had several reports of
>>>>> FreeSWITCH success stories. We will be providing more information about
>>>>> those in upcoming stories on our Web site. We've got people using
>>>>> FreeSWITCH
>>>>> in various situations as well as software developers who've added
>>>>> support
>>>>> for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues to
>>>>> grow and flourish! Thank you all for being a part of it.
>>>>>
>>>>> Take care and have a great week!
>>>>>
>>>>> --
>>>>> Michael S Collins
>>>>> ClueCon Team
>>>>> http://www.cluecon.com
>>>>> 877-7-4ACLUE
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Kristian Kielhofner
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Michael S Collins
>>> Twitter: @mercutioviz
>>> http://www.FreeSWITCH.org
>>> http://www.ClueCon.com
>>> http://www.OSTAG.org
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-dev mailing list
>>> FreeSWITCH-dev at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>> http://www.freeswitch.org
>>>
>>
>>
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org



-- 
Kristian Kielhofner


From cayliffe at gmail.com  Tue Dec 11 01:56:44 2012
From: cayliffe at gmail.com (Craig Ayliffe)
Date: Tue, 11 Dec 2012 09:56:44 +1100
Subject: [Freeswitch-dev] How to find out what changed in a release
Message-ID: <CAJMgsgtZVLqqh4Oja8FkMhPrksN+3J5tqEZM=JW_sducK0d7Cg@mail.gmail.com>

Hi,

I am looking at the ChangeLog in the tarball for the latest release
1.2.5.3, and it doesn't have an entry.
Plus I also noticed the last two entries (1.2.5.2 and 1.2.5.1) in the
ChangeLog also just say 'Maintenance release/bug fixes'.

So how would I find out what has changed, i.e. which bugs have been fixed
in each release?

Regards,

-- 
Craig
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From krice at freeswitch.org  Tue Dec 11 02:11:52 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 10 Dec 2012 17:11:52 -0600
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
In-Reply-To: <CAKDfjgfyLaAJm80dZugiYNBqBzdTBG_jsh=gT17TfCcVXWnkaA@mail.gmail.com>
Message-ID: <CCEBC858.61D67%krice@freeswitch.org>

Its the caching on the CDN... It drives me insane sometimes... Adding
something else to the list of things


On 12/10/12 4:52 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:

> 1.2.5.3 works for me, 1.2.5.2 does not (but I guess it doesn't matter
> much anymore).
> 
> On Mon, Dec 10, 2012 at 5:44 PM, Ken Rice <krice at freeswitch.org> wrote:
>> http://files.freeswitch.org/freeswitch-1.2.5.3.tar.bz2  not working?
>> 
>> Ok the CDN is starting to get on my nerves
>> 
>> K
>> 
>> 
>> On 12/10/12 4:08 PM, "Kristian Kielhofner" <kris at kriskinc.com> wrote:
>> 
>>> 404 for me
>>> 
>>> On Mon, Dec 10, 2012 at 4:12 PM, Michael Collins <msc at freeswitch.org> wrote:
>>>> Actually, Mr. Rice has said that he would get them up on the list. However,
>>>> I think if you actually click the link it will grab the file. At least it
>>>> did for me about 10 seconds ago...
>>>> 
>>>> -MC
>>>> 
>>>> 
>>>> On Mon, Dec 10, 2012 at 12:55 PM, Kristian Kielhofner <kris at kriskinc.com>
>>>> wrote:
>>>>> 
>>>>> FYI 1.2.5.2 and 1.2.5.3 still aren't available on files.freeswitch.org :(
>>>>> 
>>>>> On Mon, Dec 10, 2012 at 3:29 PM, Michael Collins <marketing at cluecon.com>
>>>>> wrote:
>>>>>> Greetings!
>>>>>> 
>>>>>> We are glad to report that the FreeSWITCH team has tagged version
>>>>>> 1.2.5.3.
>>>>>> You can download the tarball here. Anyone using 1.2.5.x should update as
>>>>>> soon as possible. We appreciate all those who have helped us with
>>>>>> testing
>>>>>> and tracking down some sneaky and pernicious little bugs.
>>>>>> 
>>>>>> On last week's conference call we spent some time talking about the XML
>>>>>> parser and some of its pre-processor directives. We discussed
>>>>>> specifically
>>>>>> how you can use the "exec" command to execute a shell script in the
>>>>>> middle
>>>>>> of XML processing. We also discussed a few tricks on how to look at the
>>>>>> source code when you need to learn about some FreeSWITCH functionality
>>>>>> that
>>>>>> otherwise is not documented. This week's conference call subject is
>>>>>> still
>>>>>> pending, so stay tuned!
>>>>>> 
>>>>>> One other item I'd like to mention is that we've had several reports of
>>>>>> FreeSWITCH success stories. We will be providing more information about
>>>>>> those in upcoming stories on our Web site. We've got people using
>>>>>> FreeSWITCH
>>>>>> in various situations as well as software developers who've added
>>>>>> support
>>>>>> for FreeSWITCH to their offerings. The FreeSWITCH ecosystem continues to
>>>>>> grow and flourish! Thank you all for being a part of it.
>>>>>> 
>>>>>> Take care and have a great week!
>>>>>> 
>>>>>> --
>>>>>> Michael S Collins
>>>>>> ClueCon Team
>>>>>> http://www.cluecon.com
>>>>>> 877-7-4ACLUE
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.com
>>>>>> 
>>>>>> FreeSWITCH-dev mailing list
>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> --
>>>>> Kristian Kielhofner
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>> 
>>>>> FreeSWITCH-dev mailing list
>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> 
>>>> --
>>>> Michael S Collins
>>>> Twitter: @mercutioviz
>>>> http://www.FreeSWITCH.org
>>>> http://www.ClueCon.com
>>>> http://www.OSTAG.org
>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>> 
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> 
>> 
>> --
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #freeswitch
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
> 
> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From msc at freeswitch.org  Tue Dec 11 02:22:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Dec 2012 15:22:45 -0800
Subject: [Freeswitch-dev] How to find out what changed in a release
In-Reply-To: <CAJMgsgtZVLqqh4Oja8FkMhPrksN+3J5tqEZM=JW_sducK0d7Cg@mail.gmail.com>
References: <CAJMgsgtZVLqqh4Oja8FkMhPrksN+3J5tqEZM=JW_sducK0d7Cg@mail.gmail.com>
Message-ID: <CAKzWOxUtzFZJ_kPraz67n4D9taf3i8yVYFEA7nNcWfYG=5MDsg@mail.gmail.com>

I would use Fisheye for this. The devs are really good at using the
"--resolved FS-xxxx" syntax when doing git commits. Look at the commits
leading up to 1.2.5.3 being tagged and you'll see which bugs specifically
were addressed.

-MC

On Mon, Dec 10, 2012 at 2:56 PM, Craig Ayliffe <cayliffe at gmail.com> wrote:

> Hi,
>
> I am looking at the ChangeLog in the tarball for the latest release
> 1.2.5.3, and it doesn't have an entry.
> Plus I also noticed the last two entries (1.2.5.2 and 1.2.5.1) in the
> ChangeLog also just say 'Maintenance release/bug fixes'.
>
> So how would I find out what has changed, i.e. which bugs have been fixed
> in each release?
>
> Regards,
>
> --
> Craig
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From william.suffill at gmail.com  Wed Dec 12 00:31:40 2012
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 11 Dec 2012 16:31:40 -0500
Subject: [Freeswitch-dev] How to find out what changed in a release
In-Reply-To: <CAKzWOxUtzFZJ_kPraz67n4D9taf3i8yVYFEA7nNcWfYG=5MDsg@mail.gmail.com>
References: <CAJMgsgtZVLqqh4Oja8FkMhPrksN+3J5tqEZM=JW_sducK0d7Cg@mail.gmail.com>
	<CAKzWOxUtzFZJ_kPraz67n4D9taf3i8yVYFEA7nNcWfYG=5MDsg@mail.gmail.com>
Message-ID: <CAF-ZooRocqNvSR5kJ9HStcx30mm+hyfFVzxPudkzM3VKMow7Nw@mail.gmail.com>

If you have the source tree checked out with git, git log can be used to
print what happened between tags as well. It's been something I been
meaning to revisit but never seem to get to. For tarball releases, Fisheye
like suggested above would be the easiest.


On Mon, Dec 10, 2012 at 6:22 PM, Michael Collins <msc at freeswitch.org> wrote:

> I would use Fisheye for this. The devs are really good at using the
> "--resolved FS-xxxx" syntax when doing git commits. Look at the commits
> leading up to 1.2.5.3 being tagged and you'll see which bugs specifically
> were addressed.
>
> -MC
>
> On Mon, Dec 10, 2012 at 2:56 PM, Craig Ayliffe <cayliffe at gmail.com> wrote:
>
>> Hi,
>>
>> I am looking at the ChangeLog in the tarball for the latest release
>> 1.2.5.3, and it doesn't have an entry.
>> Plus I also noticed the last two entries (1.2.5.2 and 1.2.5.1) in the
>> ChangeLog also just say 'Maintenance release/bug fixes'.
>>
>> So how would I find out what has changed, i.e. which bugs have been fixed
>> in each release?
>>
>> Regards,
>>
>> --
>> Craig
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Dec 12 19:54:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Dec 2012 08:54:31 -0800
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxXbXubRj4id1ND0ODUgf-8jUQT4fJxBy2RYHLxo_Dqegg@mail.gmail.com>

Hello community!

Today's conference call agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12

We will be doing a community scrum today. If he's available, Dave Kompel
(IRC: drk__) will be talking about techniques for gathering data when
FreeSWITCH crashes when running under Windows. Talk to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From dxj19831029 at gmail.com  Thu Dec 13 12:44:09 2012
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Thu, 13 Dec 2012 17:44:09 +0800
Subject: [Freeswitch-dev]  Stereo support?
Message-ID: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>

hey guys,


Have anyone try to implement stereo support in freeswitch to streaming the
music?

I tried, but only 1 channel is hardcoded inside freeswitch code everywhere,
i can't make it work properly, just wonder if any1 tried before?

Can we share some experiences?


What I tried:

File IO needs to support 2 channels
shout module needs to support 2 channels
L16 needs to support 2 channels
it should auto convert between 1 channel codec and 2 channel codec for
encoding/decoding.

The problems I had are:

* some random RTP sending error. (using PCMA codec, before encode, i did
convert audio from stereo into mono).

* music voice becomes noise. ( I tried to use MPG123_FORCE_STEREO to
convert 1 channel music to 2 channel music).



Cheers
Xijing
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From richard.screene at netdev.co.uk  Thu Dec 13 12:50:40 2012
From: richard.screene at netdev.co.uk (Richard Screene)
Date: Thu, 13 Dec 2012 09:50:40 +0000
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
References: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
Message-ID: <1DAA6857-E47C-42FE-97BB-F0C48F4E2ED0@netdev.co.uk>

bugger?. sounds complicated

On 13 Dec 2012, at 09:44, Xijing Dai wrote:

> hey guys,
> 
> 
> Have anyone try to implement stereo support in freeswitch to streaming the music?
> 
> I tried, but only 1 channel is hardcoded inside freeswitch code everywhere, i can't make it work properly, just wonder if any1 tried before?
> 
> Can we share some experiences?
> 
> 
> What I tried:
> 
> File IO needs to support 2 channels
> shout module needs to support 2 channels
> L16 needs to support 2 channels
> it should auto convert between 1 channel codec and 2 channel codec for encoding/decoding.
> 
> The problems I had are:
> 
> * some random RTP sending error. (using PCMA codec, before encode, i did convert audio from stereo into mono).
> 
> * music voice becomes noise. ( I tried to use MPG123_FORCE_STEREO to convert 1 channel music to 2 channel music).
> 
> 
> 
> Cheers
> Xijing
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

Richard Screene
Senior Developer
Drum Web Meetings 
+44 1273 936125
www.thisisdrum.com
Drum is the collaboration solution by NetDev Ltd.
Registered in England and Wales 
Company Number 04741258

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From rick at openfortress.nl  Thu Dec 13 13:01:16 2012
From: rick at openfortress.nl (Rick van Rein)
Date: Thu, 13 Dec 2012 10:01:16 +0000
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
References: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
Message-ID: <20121213100116.GB15047@newphantom.local>

Hey,

> Have anyone try to implement stereo support in freeswitch to streaming the
> music?

It'd also be a great asset to conference calls if we could hear the people
"in the room" each talk from their position.  Add stereo microphones and it
is even possible to "look at people" and direct speech more loudly their
way.

This is more sensational than you can imagine :- I once ate in a restaurant
where the situation of being blind was simulated through absolute darkness.
I was struck by how instinctively the mind registers if someone is looking at
you while speaking, almost as if it has once been useful for our defense or so.

I think this sensation could make an incredible impact on the concept of a
conference call, and make it come *much* closer to a live conversation.

> I tried, but only 1 channel is hardcoded inside freeswitch code everywhere,
> i can't make it work properly, just wonder if any1 tried before?

Can't answer that, just motivate it as a more generally useful idea.


Ciao,
 -Rick


From dxj19831029 at gmail.com  Thu Dec 13 18:18:11 2012
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Thu, 13 Dec 2012 23:18:11 +0800
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <20121213100116.GB15047@newphantom.local>
References: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
	<20121213100116.GB15047@newphantom.local>
Message-ID: <CAMsCbX_v9zSdMLvwy9GnGJk=LULzd5Dnc93bk3W2G-H=MLezJA@mail.gmail.com>

On Thu, Dec 13, 2012 at 6:01 PM, Rick van Rein <rick at openfortress.nl> wrote:

> Hey,
>
> > Have anyone try to implement stereo support in freeswitch to streaming
> the
> > music?
>
> It'd also be a great asset to conference calls if we could hear the people
> "in the room" each talk from their position.  Add stereo microphones and it
> is even possible to "look at people" and direct speech more loudly their
> way.
>
> This is more sensational than you can imagine :- I once ate in a restaurant
> where the situation of being blind was simulated through absolute darkness.
> I was struck by how instinctively the mind registers if someone is looking
> at
> you while speaking, almost as if it has once been useful for our defense
> or so.
>
> I think this sensation could make an incredible impact on the concept of a
> conference call, and make it come *much* closer to a live conversation.
>

Great idea. I did have a look at conference algorithm. It's first added up
all mic
voice wave and than minus listener's voice from it. By this mean, we could
also
increase volume of voice of whom looked at listener and added up before
sent the wav to listeners.

However, the difficult part is how to figure out if someone is looked at
someone?



> I tried, but only 1 channel is hardcoded inside freeswitch code
everywhere,
> i can't make it work properly, just wonder if any1 tried before?

Can't answer that, just motivate it as a more generally useful idea.
>
>
> Ciao,
>  -Rick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Thu Dec 13 19:20:46 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 10:20:46 -0600
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <CAMsCbX_v9zSdMLvwy9GnGJk=LULzd5Dnc93bk3W2G-H=MLezJA@mail.gmail.com>
References: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
	<20121213100116.GB15047@newphantom.local>
	<CAMsCbX_v9zSdMLvwy9GnGJk=LULzd5Dnc93bk3W2G-H=MLezJA@mail.gmail.com>
Message-ID: <CAKbxfG_nwrH6A+DwnEeGaZbds-HjSO0mPvphBuVQfiC7_q_9gQ@mail.gmail.com>

We have the plumbing for it but we did not currently attempt to support it
yet.
The codecs and file handles all support a number of channels.  Basically
all the places where we detect sample rate differences and engage
transcoding we would need to mux/demux stereo/mono.

For the conference, my idea was to make it so you can create many channel
conferences and either call in with stereo RTP on opus or some mono codec
into one specific channel like you could call 2 mono speakerphones into 2
separate channels of the conference on the same table or something.





On Thu, Dec 13, 2012 at 9:18 AM, Xijing Dai <dxj19831029 at gmail.com> wrote:

>
>
> On Thu, Dec 13, 2012 at 6:01 PM, Rick van Rein <rick at openfortress.nl>wrote:
>
>> Hey,
>>
>> > Have anyone try to implement stereo support in freeswitch to streaming
>> the
>> > music?
>>
>> It'd also be a great asset to conference calls if we could hear the people
>> "in the room" each talk from their position.  Add stereo microphones and
>> it
>> is even possible to "look at people" and direct speech more loudly their
>> way.
>>
>> This is more sensational than you can imagine :- I once ate in a
>> restaurant
>> where the situation of being blind was simulated through absolute
>> darkness.
>> I was struck by how instinctively the mind registers if someone is
>> looking at
>> you while speaking, almost as if it has once been useful for our defense
>> or so.
>>
>> I think this sensation could make an incredible impact on the concept of a
>> conference call, and make it come *much* closer to a live conversation.
>>
>
> Great idea. I did have a look at conference algorithm. It's first added up
> all mic
> voice wave and than minus listener's voice from it. By this mean, we could
> also
> increase volume of voice of whom looked at listener and added up before
> sent the wav to listeners.
>
> However, the difficult part is how to figure out if someone is looked at
> someone?
>
>
>
>
> > I tried, but only 1 channel is hardcoded inside freeswitch code
> everywhere,
> > i can't make it work properly, just wonder if any1 tried before?
>
> Can't answer that, just motivate it as a more generally useful idea.
>>
>>
>> Ciao,
>>  -Rick
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From krice at freeswitch.org  Thu Dec 13 19:54:13 2012
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 13 Dec 2012 10:54:13 -0600
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <20121213100116.GB15047@newphantom.local>
Message-ID: <CCEF6455.61F8E%krice@freeswitch.org>

Maybe use 2 conference bridges and a SIP UA modified to make 2 calls
1000L at conference.bridge  1000R at conference.bridge

Left channel @ 1000L, right channel @ 1000R, using other conference magic
you could even steer people around the room by playing with their L and R
gains

K



On 12/13/12 4:01 AM, "Rick van Rein" <rick at openfortress.nl> wrote:

> Hey,
> 
>> Have anyone try to implement stereo support in freeswitch to streaming the
>> music?
> 
> It'd also be a great asset to conference calls if we could hear the people
> "in the room" each talk from their position.  Add stereo microphones and it
> is even possible to "look at people" and direct speech more loudly their
> way.
> 
> This is more sensational than you can imagine :- I once ate in a restaurant
> where the situation of being blind was simulated through absolute darkness.
> I was struck by how instinctively the mind registers if someone is looking at
> you while speaking, almost as if it has once been useful for our defense or
> so.
> 
> I think this sensation could make an incredible impact on the concept of a
> conference call, and make it come *much* closer to a live conversation.
> 
>> I tried, but only 1 channel is hardcoded inside freeswitch code everywhere,
>> i can't make it work properly, just wonder if any1 tried before?
> 
> Can't answer that, just motivate it as a more generally useful idea.
> 
> 
> Ciao,
>  -Rick
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From rick at openfortress.nl  Fri Dec 14 02:07:07 2012
From: rick at openfortress.nl (Rick van Rein)
Date: Thu, 13 Dec 2012 23:07:07 +0000
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <CAMsCbX_v9zSdMLvwy9GnGJk=LULzd5Dnc93bk3W2G-H=MLezJA@mail.gmail.com>
References: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
	<20121213100116.GB15047@newphantom.local>
	<CAMsCbX_v9zSdMLvwy9GnGJk=LULzd5Dnc93bk3W2G-H=MLezJA@mail.gmail.com>
Message-ID: <20121213230707.GE15047@newphantom.local>

Hello,

> the difficult part is how to figure out if someone is looked at
> someone?

Assuming that they delivered stereo microphone sound, which is a stretch in
today's setups (a chicken-egg problem) it would be trivial I think.  You
would basically do a calculation like

	tan (alpha) = (R - L) / 0.5 * (R + L)

and find in alpha the angle right from straight-forward.  Of course it is
useful to dampen movements with a low-pass filter on momentary calculations,
and/or to make these calculations over periods rather than for each
individual audio sample pair.

If the switch is aware what person is put in what position it could figure
out which other is addressed.  That might call for a switch that does this
work separately for each participant.

I suppose it would be practical to imagine conference partners sitting in a
circle.  N persons would be spread with 360 degrees / N apart.  If they
spoke out under an angle alpha, than their i'th neighbour would hear the
sound at an angle (alpha + i*360/N degrees).  And that angle would be used
in the opposite way as the tangent-formula above.

Sorry about the maths... I think it helps to clarify the approach though.
If you'd like me to, I could try to find an elegant and *practical* way of
modelling this with a simple matrix operation.  Let me know if you'd like
me to contribute that.  It'd mean that mixing inputs and forming outputs
from them is a nice operation with 2N inputs and 2N outputs with all the
above isolated into a series of multiplications and additions.  Of the sort
that would easily be met in hardware optimisations, if present.

I'm sure that hardware vendors won't mind adding a layer of luxoury models
for telephony... certainly not for business applications where they money
is available... but AFAIK there are no stereo switches yet.  Imagine a
good-quality codec (screwing ISDN, embracing OPUS) and the advantages of
stereo communication, we could have a good laugh at all those people who
still think 3000 Hz, audio-only and IPv4 define the VoIP arena.


Cheers,
 -Rick


From anthony.minessale at gmail.com  Fri Dec 14 03:17:06 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Dec 2012 18:17:06 -0600
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <20121213230707.GE15047@newphantom.local>
References: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
	<20121213100116.GB15047@newphantom.local>
	<CAMsCbX_v9zSdMLvwy9GnGJk=LULzd5Dnc93bk3W2G-H=MLezJA@mail.gmail.com>
	<20121213230707.GE15047@newphantom.local>
Message-ID: <CAKbxfG8yFZrL8ozX_68E9WBZ0uH5M1aEf5vMTcx5tvH3Xx1Nxw@mail.gmail.com>

If everyone develops an interest in it and we can finde some equipment that
actually will take advantage of it then I am all for it.  I've just been
waiting for that to happen.  It does seem like that's pretty soon.....


On Thu, Dec 13, 2012 at 5:07 PM, Rick van Rein <rick at openfortress.nl> wrote:

> Hello,
>
> > the difficult part is how to figure out if someone is looked at
> > someone?
>
> Assuming that they delivered stereo microphone sound, which is a stretch in
> today's setups (a chicken-egg problem) it would be trivial I think.  You
> would basically do a calculation like
>
>         tan (alpha) = (R - L) / 0.5 * (R + L)
>
> and find in alpha the angle right from straight-forward.  Of course it is
> useful to dampen movements with a low-pass filter on momentary
> calculations,
> and/or to make these calculations over periods rather than for each
> individual audio sample pair.
>
> If the switch is aware what person is put in what position it could figure
> out which other is addressed.  That might call for a switch that does this
> work separately for each participant.
>
> I suppose it would be practical to imagine conference partners sitting in a
> circle.  N persons would be spread with 360 degrees / N apart.  If they
> spoke out under an angle alpha, than their i'th neighbour would hear the
> sound at an angle (alpha + i*360/N degrees).  And that angle would be used
> in the opposite way as the tangent-formula above.
>
> Sorry about the maths... I think it helps to clarify the approach though.
> If you'd like me to, I could try to find an elegant and *practical* way of
> modelling this with a simple matrix operation.  Let me know if you'd like
> me to contribute that.  It'd mean that mixing inputs and forming outputs
> from them is a nice operation with 2N inputs and 2N outputs with all the
> above isolated into a series of multiplications and additions.  Of the sort
> that would easily be met in hardware optimisations, if present.
>
> I'm sure that hardware vendors won't mind adding a layer of luxoury models
> for telephony... certainly not for business applications where they money
> is available... but AFAIK there are no stereo switches yet.  Imagine a
> good-quality codec (screwing ISDN, embracing OPUS) and the advantages of
> stereo communication, we could have a good laugh at all those people who
> still think 3000 Hz, audio-only and IPv4 define the VoIP arena.
>
>
> Cheers,
>  -Rick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From steveayre at gmail.com  Fri Dec 14 14:17:54 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 14 Dec 2012 11:17:54 +0000
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <20121213230707.GE15047@newphantom.local>
References: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
	<20121213100116.GB15047@newphantom.local>
	<CAMsCbX_v9zSdMLvwy9GnGJk=LULzd5Dnc93bk3W2G-H=MLezJA@mail.gmail.com>
	<20121213230707.GE15047@newphantom.local>
Message-ID: <CAFiqYukqo6Eq2B2RLKWmQmmZJ6y7NcpAtxryECm89btVamP8+w@mail.gmail.com>

Bear in mind that doing this, you'll need to do the calculation for every
listening member. So instead of just muxing the audio for every speaker
once and broadcasting it to all members, you'll have to do that for every
single user. So much more work on the CPU, especially as your conferences
grow.

So you'd want to keep the traditional mod_conference working the way it is
now, and have any stereo support like this in something
like mod_conference_spatial (I'm guessing the code would diverge too much
to make it a single config option). It'd be a cool feature, but you'd want
it opt-in.

-Steve




On 13 December 2012 23:07, Rick van Rein <rick at openfortress.nl> wrote:

> Hello,
>
> > the difficult part is how to figure out if someone is looked at
> > someone?
>
> Assuming that they delivered stereo microphone sound, which is a stretch in
> today's setups (a chicken-egg problem) it would be trivial I think.  You
> would basically do a calculation like
>
>         tan (alpha) = (R - L) / 0.5 * (R + L)
>
> and find in alpha the angle right from straight-forward.  Of course it is
> useful to dampen movements with a low-pass filter on momentary
> calculations,
> and/or to make these calculations over periods rather than for each
> individual audio sample pair.
>
> If the switch is aware what person is put in what position it could figure
> out which other is addressed.  That might call for a switch that does this
> work separately for each participant.
>
> I suppose it would be practical to imagine conference partners sitting in a
> circle.  N persons would be spread with 360 degrees / N apart.  If they
> spoke out under an angle alpha, than their i'th neighbour would hear the
> sound at an angle (alpha + i*360/N degrees).  And that angle would be used
> in the opposite way as the tangent-formula above.
>
> Sorry about the maths... I think it helps to clarify the approach though.
> If you'd like me to, I could try to find an elegant and *practical* way of
> modelling this with a simple matrix operation.  Let me know if you'd like
> me to contribute that.  It'd mean that mixing inputs and forming outputs
> from them is a nice operation with 2N inputs and 2N outputs with all the
> above isolated into a series of multiplications and additions.  Of the sort
> that would easily be met in hardware optimisations, if present.
>
> I'm sure that hardware vendors won't mind adding a layer of luxoury models
> for telephony... certainly not for business applications where they money
> is available... but AFAIK there are no stereo switches yet.  Imagine a
> good-quality codec (screwing ISDN, embracing OPUS) and the advantages of
> stereo communication, we could have a good laugh at all those people who
> still think 3000 Hz, audio-only and IPv4 define the VoIP arena.
>
>
> Cheers,
>  -Rick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From rick at openfortress.nl  Fri Dec 14 11:08:03 2012
From: rick at openfortress.nl (Rick van Rein)
Date: Fri, 14 Dec 2012 08:08:03 +0000
Subject: [Freeswitch-dev] Stereo support?
In-Reply-To: <CAKbxfG8yFZrL8ozX_68E9WBZ0uH5M1aEf5vMTcx5tvH3Xx1Nxw@mail.gmail.com>
References: <CAMsCbX9SG_MDN00dWyy5AmMYtgNf3yhen4tQTJgujeT6DQ5i5w@mail.gmail.com>
	<20121213100116.GB15047@newphantom.local>
	<CAMsCbX_v9zSdMLvwy9GnGJk=LULzd5Dnc93bk3W2G-H=MLezJA@mail.gmail.com>
	<20121213230707.GE15047@newphantom.local>
	<CAKbxfG8yFZrL8ozX_68E9WBZ0uH5M1aEf5vMTcx5tvH3Xx1Nxw@mail.gmail.com>
Message-ID: <20121214080803.GC18943@newphantom.local>

Hey,

> If everyone develops an interest in it and we can finde some equipment that
> actually will take advantage of it then I am all for it.  I've just been
> waiting for that to happen.  It does seem like that's pretty soon.....

Just like me!

I'm working on open source SIP firmware,

http://0cpm.org/firmerware/

and have also included this as a remark in its documentation, to tickle
hardware manufacturers,

https://github.com/vanrein/0cpm-Firmerware/blob/master/doc/hardware-design.rst

I very much liked your idea of using two speakerphones BTW!  To fill a
circle, we'd probably need three channels, but it might work out fine
with two and spreading people over half a circle.


Coming back to my experience of someone talking directly to you when
vision is blocked -- not sure what causes that exactlye.  The stereo
represents just the angle from which the sound comes, but the info
that makes you feel that the sound is not bouncing indirectly to
you might be something else.  Not sure here, probably blind people
know more about it.


Cheers,
 -Rick


From everydayuser11 at gmail.com  Sat Dec 15 02:18:18 2012
From: everydayuser11 at gmail.com (Everyday User)
Date: Fri, 14 Dec 2012 15:18:18 -0800
Subject: [Freeswitch-dev] Best way to test DID is hitting Freeswitch
Message-ID: <CAAbPOc68FDMXiy41TDM8VE1WdDPukZr8BePU_pZaE8+ECzokRg@mail.gmail.com>

I have thousands of DID's and wondering what is the best method to run
tests to see if these numbers are hitting our Freeswitch server.  We have
problems with our carrier assigning our numbers to other companies and need
to preemptively know if the number is not pointing to us anymore before our
clients find out.

Currently, I run a command to originate a call for each DID which updates a
database entry to see if that number went through are server.  I am looking
for a more efficient way to test all these numbers. Thanks.

- Jim
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From krice at freeswitch.org  Sun Dec 16 01:00:15 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 15 Dec 2012 16:00:15 -0600
Subject: [Freeswitch-dev] Best way to test DID is hitting Freeswitch
In-Reply-To: <CAAbPOc68FDMXiy41TDM8VE1WdDPukZr8BePU_pZaE8+ECzokRg@mail.gmail.com>
Message-ID: <CCF24F0F.62120%krice@freeswitch.org>

That?s about the only way you can tell that... I have to question your
choice of carrier if they are doing that on a routine basis...


On 12/14/12 5:18 PM, "Everyday User" <everydayuser11 at gmail.com> wrote:

> I have thousands of DID's and wondering what is the best method to run tests
> to see if these numbers are hitting our Freeswitch server. ?We have problems
> with our carrier assigning our numbers to other companies and need to
> preemptively know if the number is not pointing to us anymore before our
> clients find out.
> 
> Currently, I run a command to originate a call for each DID which updates a
> database entry to see if that number went through are server. ?I am looking
> for a more efficient way to test all these numbers. Thanks.
> 
> - Jim
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From steveayre at gmail.com  Sun Dec 16 04:56:11 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 16 Dec 2012 01:56:11 +0000
Subject: [Freeswitch-dev] Best way to test DID is hitting Freeswitch
In-Reply-To: <CCF24F0F.62120%krice@freeswitch.org>
References: <CAAbPOc68FDMXiy41TDM8VE1WdDPukZr8BePU_pZaE8+ECzokRg@mail.gmail.com>
	<CCF24F0F.62120%krice@freeswitch.org>
Message-ID: <CAFiqYuk6y7UxGw0um=XwYRZzMQ=CFy+TrXTy1HVJetY0T3MTFw@mail.gmail.com>

Your other option is if you're storing CDRs (which you should be) is to
have some tool analysing your calls against a list of DIDs to flag up which
ones are not receiving any calls. Then you can check those. At the very
least that might decrease the number of originates you're having to do.

No calls might mean just no calls, but it might also flag up that the DID
has been reassigned to point elsewhere.

But I agree with Ken... they shouldn't be doing this, and if it's becoming
the norm I'd look elsewhere.



On 15 December 2012 22:00, Ken Rice <krice at freeswitch.org> wrote:

>  That?s about the only way you can tell that... I have to question your
> choice of carrier if they are doing that on a routine basis...
>
>
> On 12/14/12 5:18 PM, "Everyday User" <everydayuser11 at gmail.com> wrote:
>
> I have thousands of DID's and wondering what is the best method to run
> tests to see if these numbers are hitting our Freeswitch server.  We have
> problems with our carrier assigning our numbers to other companies and need
> to preemptively know if the number is not pointing to us anymore before our
> clients find out.
>
> Currently, I run a command to originate a call for each DID which updates
> a database entry to see if that number went through are server.  I am
> looking for a more efficient way to test all these numbers. Thanks.
>
> - Jim
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
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From hagai.sela at gmail.com  Sun Dec 16 22:30:40 2012
From: hagai.sela at gmail.com (hagai sela)
Date: Sun, 16 Dec 2012 21:30:40 +0200
Subject: [Freeswitch-dev] app that sends and receives faxes
Message-ID: <CAOKbVndsvJzLpjR2ntDSxQWu-=g4DUFFbbUBE2sUz5FiK+b0cA@mail.gmail.com>

Hi,
I want to write a small C# app that sends and receives faxes over
SIP/RTP/T.38 and I thought about using FreeSwitch.
I tried to look for examples on this and couldn't find any. Can anyone post
such an example?

Thanks.
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From krice at freeswitch.org  Mon Dec 17 01:42:35 2012
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 16 Dec 2012 16:42:35 -0600
Subject: [Freeswitch-dev] app that sends and receives faxes
In-Reply-To: <CAOKbVndsvJzLpjR2ntDSxQWu-=g4DUFFbbUBE2sUz5FiK+b0cA@mail.gmail.com>
Message-ID: <CCF3AA7B.6230D%krice@freeswitch.org>

You didn?t look very hard, freeswitch does this out of the box...  Just note
T.38 does not work over RTP, it uses UPDTL for a transport...

Just google freeswitch t.38 or freeswitch fax


On 12/16/12 1:30 PM, "hagai sela" <hagai.sela at gmail.com> wrote:

> Hi,
> I want to write a small C# app that sends and receives faxes over SIP/RTP/T.38
> and I thought about using FreeSwitch.
> I tried to look for examples on this and couldn't find any. Can anyone post
> such an example?
> 
> Thanks.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From kris at kriskinc.com  Mon Dec 17 02:22:15 2012
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Sun, 16 Dec 2012 18:22:15 -0500
Subject: [Freeswitch-dev] Best way to test DID is hitting Freeswitch
In-Reply-To: <CAAbPOc68FDMXiy41TDM8VE1WdDPukZr8BePU_pZaE8+ECzokRg@mail.gmail.com>
References: <CAAbPOc68FDMXiy41TDM8VE1WdDPukZr8BePU_pZaE8+ECzokRg@mail.gmail.com>
Message-ID: <CAKDfjgf0EG8NT8QQpdZ=JeA3YC7dXUyLDwnFmc5S=JWy1EDRng@mail.gmail.com>

Buy a circuit/minutes/whatever from another carrier.

Write a match at the top of your dialplan to catch calls coming from a
number assigned to the circuit above.  Make the match conditional for
only the caller id number of that circuit (matching any destination
number).  Because the caller id number is globally unique you'll be
the only person using it.

Then use FreeSWITCH originate commands to send calls to the DIDs
you're porting on a periodic basis from that number.

When you see the call come back and hit your server you'll know the
number has been ported to you.

Make sure to take care and not harass your customers but I'll leave
that part of the exercise up to you.

On Fri, Dec 14, 2012 at 6:18 PM, Everyday User <everydayuser11 at gmail.com> wrote:
> I have thousands of DID's and wondering what is the best method to run tests
> to see if these numbers are hitting our Freeswitch server.  We have problems
> with our carrier assigning our numbers to other companies and need to
> preemptively know if the number is not pointing to us anymore before our
> clients find out.
>
> Currently, I run a command to originate a call for each DID which updates a
> database entry to see if that number went through are server.  I am looking
> for a more efficient way to test all these numbers. Thanks.
>
> - Jim
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From dave at 3c.co.uk  Mon Dec 17 02:50:54 2012
From: dave at 3c.co.uk (David Knell)
Date: Sun, 16 Dec 2012 23:50:54 -0000
Subject: [Freeswitch-dev] Best way to test DID is hitting Freeswitch
In-Reply-To: <CAKDfjgf0EG8NT8QQpdZ=JeA3YC7dXUyLDwnFmc5S=JWy1EDRng@mail.gmail.com>
References: <CAAbPOc68FDMXiy41TDM8VE1WdDPukZr8BePU_pZaE8+ECzokRg@mail.gmail.com>
	<CAKDfjgf0EG8NT8QQpdZ=JeA3YC7dXUyLDwnFmc5S=JWy1EDRng@mail.gmail.com>
Message-ID: <024201cddbe8$303863f0$90a92bd0$@co.uk>

This - exactly this - use a carrier for your test calls that's
distinct from those who deliver calls to you.

--Dave

-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: 16 December 2012 23:22
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Best way to test DID is hitting Freeswitch

Buy a circuit/minutes/whatever from another carrier.

Write a match at the top of your dialplan to catch calls coming from a
number assigned to the circuit above.  Make the match conditional for
only the caller id number of that circuit (matching any destination
number).  Because the caller id number is globally unique you'll be
the only person using it.

Then use FreeSWITCH originate commands to send calls to the DIDs
you're porting on a periodic basis from that number.

When you see the call come back and hit your server you'll know the
number has been ported to you.

Make sure to take care and not harass your customers but I'll leave
that part of the exercise up to you.

On Fri, Dec 14, 2012 at 6:18 PM, Everyday User <everydayuser11 at gmail.com>
wrote:
> I have thousands of DID's and wondering what is the best method to run
tests
> to see if these numbers are hitting our Freeswitch server.  We have
problems
> with our carrier assigning our numbers to other companies and need to
> preemptively know if the number is not pointing to us anymore before our
> clients find out.
>
> Currently, I run a command to originate a call for each DID which updates
a
> database entry to see if that number went through are server.  I am
looking
> for a more efficient way to test all these numbers. Thanks.
>
> - Jim
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org




From anton.jugatsu at gmail.com  Mon Dec 17 06:10:38 2012
From: anton.jugatsu at gmail.com (Anton Kvashenkin)
Date: Mon, 17 Dec 2012 07:10:38 +0400
Subject: [Freeswitch-dev] Best way to test DID is hitting Freeswitch
In-Reply-To: <024201cddbe8$303863f0$90a92bd0$@co.uk>
References: <CAAbPOc68FDMXiy41TDM8VE1WdDPukZr8BePU_pZaE8+ECzokRg@mail.gmail.com>
	<CAKDfjgf0EG8NT8QQpdZ=JeA3YC7dXUyLDwnFmc5S=JWy1EDRng@mail.gmail.com>
	<024201cddbe8$303863f0$90a92bd0$@co.uk>
Message-ID: <CANBhyPDRMoriG2NcKO_bMomVdGPvZCtsN=7oX5--K6LOs8KLqQ@mail.gmail.com>

ngrep -d <iface> -qt -W byline 'sip: INVITE' 'port 5060 or port 5080'

:)


2012/12/17 David Knell <dave at 3c.co.uk>

> This - exactly this - use a carrier for your test calls that's
> distinct from those who deliver calls to you.
>
> --Dave
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Kristian
> Kielhofner
> Sent: 16 December 2012 23:22
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Best way to test DID is hitting Freeswitch
>
> Buy a circuit/minutes/whatever from another carrier.
>
> Write a match at the top of your dialplan to catch calls coming from a
> number assigned to the circuit above.  Make the match conditional for
> only the caller id number of that circuit (matching any destination
> number).  Because the caller id number is globally unique you'll be
> the only person using it.
>
> Then use FreeSWITCH originate commands to send calls to the DIDs
> you're porting on a periodic basis from that number.
>
> When you see the call come back and hit your server you'll know the
> number has been ported to you.
>
> Make sure to take care and not harass your customers but I'll leave
> that part of the exercise up to you.
>
> On Fri, Dec 14, 2012 at 6:18 PM, Everyday User <everydayuser11 at gmail.com>
> wrote:
> > I have thousands of DID's and wondering what is the best method to run
> tests
> > to see if these numbers are hitting our Freeswitch server.  We have
> problems
> > with our carrier assigning our numbers to other companies and need to
> > preemptively know if the number is not pointing to us anymore before our
> > clients find out.
> >
> > Currently, I run a command to originate a call for each DID which updates
> a
> > database entry to see if that number went through are server.  I am
> looking
> > for a more efficient way to test all these numbers. Thanks.
> >
> > - Jim
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> > http://www.freeswitch.org
> >
>
>
>
> --
> Kristian Kielhofner
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From rick at openfortress.nl  Mon Dec 17 10:08:18 2012
From: rick at openfortress.nl (Rick van Rein)
Date: Mon, 17 Dec 2012 07:08:18 +0000
Subject: [Freeswitch-dev] app that sends and receives faxes
In-Reply-To: <CCF3AA7B.6230D%krice@freeswitch.org>
References: <CAOKbVndsvJzLpjR2ntDSxQWu-=g4DUFFbbUBE2sUz5FiK+b0cA@mail.gmail.com>
	<CCF3AA7B.6230D%krice@freeswitch.org>
Message-ID: <20121217070818.GB31008@newphantom.local>

Hi,

> You didn?t look very hard, freeswitch does this out of the box...  Just note
> T.38 does not work over RTP, it uses UPDTL for a transport...

Actually, it can do both.

http://www.rfc-editor.org/rfc/rfc6466.txt

UDPTL was created before RTP, but has been superceded, except that it
won't die because people implement only UDPTL.  Fax implementations
should IMHO prefer to use image/t38 over RTP/UDP or RTP/TCP.  It will
mean that generic tooling is made accessible to fax.

Cheers,
 -Rick


From hagai.sela at gmail.com  Mon Dec 17 10:36:02 2012
From: hagai.sela at gmail.com (hagai)
Date: Mon, 17 Dec 2012 09:36:02 +0200
Subject: [Freeswitch-dev] app that sends and receives faxes
In-Reply-To: <CCF3AA7B.6230D%krice@freeswitch.org>
References: <CCF3AA7B.6230D%krice@freeswitch.org>
Message-ID: <1355729762.18340.4.camel@babar>

I know freeswitch does this out of the box, I just wanted an example on
how to use the libfreeswitch API to do it.

On Sun, 2012-12-16 at 16:42 -0600, Ken Rice wrote:
> You didn?t look very hard, freeswitch does this out of the box...
>  Just note T.38 does not work over RTP, it uses UPDTL for a
> transport... 
> 
> Just google freeswitch t.38 or freeswitch fax
> 
> 
> On 12/16/12 1:30 PM, "hagai sela" <hagai.sela at gmail.com> wrote:
> 
>         Hi,
>         I want to write a small C# app that sends and receives faxes
>         over SIP/RTP/T.38 and I thought about using FreeSwitch.
>         I tried to look for examples on this and couldn't find any.
>         Can anyone post such an example?
>         
>         Thanks.
>         
>         
>         ______________________________________________________________
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.com
>         
>         FreeSWITCH-dev mailing list
>         FreeSWITCH-dev at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>         http://www.freeswitch.org
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org




From krice at freeswitch.org  Mon Dec 17 17:57:18 2012
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 17 Dec 2012 08:57:18 -0600
Subject: [Freeswitch-dev] app that sends and receives faxes
In-Reply-To: <1355729762.18340.4.camel@babar>
Message-ID: <CCF48EEE.62350%krice@freeswitch.org>

I don't know if there is a good example of directly accessing the
libfreeswitch api for doing faxing... Its probably easier to just install a
stripped down version of freeswitch then control it via either
mod_eventsocket or mod_xml_rpc


On 12/17/12 1:36 AM, "hagai" <hagai.sela at gmail.com> wrote:

> I know freeswitch does this out of the box, I just wanted an example on
how to
> use the libfreeswitch API to do it.

On Sun, 2012-12-16 at 16:42 -0600, Ken
> Rice wrote:
> You didn?t look very hard, freeswitch does this out of the
> box...
>  Just note T.38 does not work over RTP, it uses UPDTL for a
>
> transport... 
> 
> Just google freeswitch t.38 or freeswitch fax
> 
> 
> On
> 12/16/12 1:30 PM, "hagai sela" <hagai.sela at gmail.com> wrote:
> 
>
> Hi,
>         I want to write a small C# app that sends and receives faxes
>
> over SIP/RTP/T.38 and I thought about using FreeSwitch.
>         I tried to
> look for examples on this and couldn't find any.
>         Can anyone post
> such an example?
>         
>         Thanks.
>         
>         
>
> ______________________________________________________________
>
> _________________________________________________________________________>
> Professional FreeSWITCH Consulting Services:
>
> consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>
> 
>         
>
> 
>         
>         Official FreeSWITCH Sites
>
> http://www.freeswitch.org
>         http://wiki.freeswitch.org
>
> http://www.cluecon.com
>         
>         FreeSWITCH-dev mailing list
>
> FreeSWITCH-dev at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>
> http://www.freeswitch.org
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
>
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
> _________________________________________________________________________>
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
>
> http://www.freeswitchsolutions.com
> 
> FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server
> 
> 
> Official FreeSWITCH Sites
>
> http://www.freeswitch.org
> http://wiki.freeswitch.org
>
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
>
> FreeSWITCH-dev at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>
> http://www.freeswitch.org



_________________________________________________
> ________________________
Professional FreeSWITCH Consulting
> Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

FreeSW
> ITCH-powered IP PBX: The CudaTel Communication
> Server


Official FreeSWITCH
> Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.
> com

FreeSWITCH-dev mailing
> list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/l
> istinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options
> /freeswitch-dev
http://www.freeswitch.org


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From marketing at cluecon.com  Mon Dec 17 22:11:12 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 17 Dec 2012 11:11:12 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVxZ==+Q=sXAZo_HhJw2m2dxx5H_3wsC3-pXJLZahDF7w@mail.gmail.com>

A happy and cold (in the northern hemisphere) Monday to you all!

I'd like to start this week's news and notes by alerting everyone to the
fact that there's been a significant addition to the functionality of the
XML dialplan. If you've been around FreeSWITCH for any length of time
you've probably read the words, "Nested conditions not allowed!" in yours
or someone else's FreeSWITCH logs. As of last
Friday<http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=9b7b46dbca9c4c458121dc80c3eeedf8e5ec9643>that
has changed! In response to Jira
FS-4935 <http://jira.freeswitch.org/browse/FS-4935> Anthony has added
provisional support for nested <condition> tags inside the dialplan.
(Thanks to IRC user vipkilla for adding this to the
wiki<http://wiki.freeswitch.org/wiki/Dialplan_XML#Nested_Conditions>already.)
As you may know we are working the second edition of the FreeSWITCH
"bridge" book<https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-telephony-systems/book>.
I will be updating chapters 5 and 8 to reflect this new change.

It seems appropriate that with this new feature we should talk about it on this
week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_19>.
Ken Rice and I will work up some simple examples of how to use the nested
conditions and how they relate to the existing XML dialplan controls such
as the break attribute and the regex tag. If you have a dialplan example
that works well with nested conditions please email Ken and me off list.

One last reminder for our Windows users: Dave Kompel shared with us some useful
information <http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_12> for
gathering debug data when FreeSWITCH crashes under Windows. Our other main
Windows guru, Jeff Lenk, was also on the call and gave some helpful input.
If you are running under Windows you now have more tools at your disposal
with which to analyze crash data and open Jira tickets.

Have a great week and we'll talk to you on Wednesday.

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From sparklezou at 163.com  Tue Dec 18 06:20:45 2012
From: sparklezou at 163.com (sparklezou)
Date: Tue, 18 Dec 2012 11:20:45 +0800
Subject: [Freeswitch-dev] About module "Mod_pandsp",
	send the fax by email
In-Reply-To: <50B3FEC6.2060709@163.com>
References: <50B3BB83.60700@gmail.com><50B3FEC6.2060709@163.com>
Message-ID: <6bfa4541.304e.13bac095d0c.Coremail.sparklezou@163.com>

Hi All,

I would like to implement the following feature on fax module "Mod_pandsp".

Fax-to-Eaml, this feature should be simular to "Voicemail-to-Email".

1. hardware fax detection, then invoke "Mod_pandsp" at FreeSwitch side. Then send the received fax to the individual email. The email info should be set in the user account.

2.software fax detection, detected by "spandsp_start_fax_detect". But it should "answer" the call firstly.

3. The caller part side wait for a fax tone manually, then start to transfer the fax. Answer the call as common. Then press "*" or "#" to send fax tone to receive fax by email.


Email-to-Fax

1. send the document which should be faxed as attachment. and in the email title, it's the destination number. 

2. Freeswitch will check the mailbox timely. send the fax. Then reply one email, success or NOT.

Thanks!

2012-12-18



sparklezou



????sparklezou
?????2012-11-27 07:52
???Re: Re: [Freeswitch-dev] About module "Mod_pandsp"
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

Correct! 

I hope the emaill function for voice-message and fax file coulde be combind.

Both "Mod_spandsp" and "Mod_voicemail" could get the usre email configuration from the user profile. such as the current "vm-enabled", "vm-mailto", "vm-mailfrom" parameters in voice-message.

It will be much easy to implement fax2email function. And much more coupling.

Most of the email fuctions should be the same for both modules.

Thanks!

2012-11-27



sparklezou



????Abaci
?????2012-11-27 02:57
???Re: [Freeswitch-dev] About module "Mod_pandsp"
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

I think what he's looking for is automatic fax to email like Voicemail to email is handled. this is not something FreeSWITCH has, not sure if it's because of related patents or something else.

On 11/26/2012 9:58 AM, sparklezou wrote:

Hi Mod_spandsp developers,

Is it possible to develop Mod_spandsp like "Mod_voicemail"?

I means, set all of the related parameters in the user configure file. such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail sender configuration, such as SMTP.

Something seemd the same, send "voice message" via eamil. Send "fax file" via email.

Thanks!

2012-11-26



sparklezou

 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org
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From jaybinks at gmail.com  Tue Dec 18 07:26:13 2012
From: jaybinks at gmail.com (jay binks)
Date: Tue, 18 Dec 2012 14:26:13 +1000
Subject: [Freeswitch-dev] About module "Mod_pandsp",
	send the fax by email
In-Reply-To: <6bfa4541.304e.13bac095d0c.Coremail.sparklezou@163.com>
References: <50B3BB83.60700@gmail.com> <50B3FEC6.2060709@163.com>
	<6bfa4541.304e.13bac095d0c.Coremail.sparklezou@163.com>
Message-ID: <CAO47HpehSFgFLmAkSp77m3nPxGGa3C5fY_0NNvxK8pM+X4Qhrw@mail.gmail.com>

im interested in your module...
"mod p and sp" sounds quite interesting, please tell me where I can join
your mailing list ?




On 18 December 2012 13:20, sparklezou <sparklezou at 163.com> wrote:

> **
> Hi All,
>
> I would like to implement the following feature on fax module "Mod_pandsp".
>
> Fax-to-Eaml, this feature should be simular to "Voicemail-to-Email".
>
> 1. hardware fax detection, then invoke "Mod_pandsp" at FreeSwitch side.
> Then send the received fax to the individual email. The email info should
> be set in the user account.
>
> 2.software fax detection, detected by "spandsp_start_fax_detect". But it
> should "answer" the call firstly.
>
> 3. The caller part side wait for a fax tone manually, then start to
> transfer the fax. Answer the call as common. Then press "*" or "#" to send
> fax tone to receive fax by email.
>
>
> Email-to-Fax
>
> 1. send the document which should be faxed as attachment. and in the email
> title, it's the destination number.
>
> 2. Freeswitch will check the mailbox timely. send the fax. Then reply one
> email, success or NOT.
>
> Thanks!
>
> 2012-12-18
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-11-27 07:52
> *???*Re: Re: [Freeswitch-dev] About module "Mod_pandsp"
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*
>
>  Hi All,
>
> Correct!
>
> I hope the emaill function for voice-message and fax file coulde be
> combind.
>
> Both "Mod_spandsp" and "Mod_voicemail" could get the usre email
> configuration from the user profile. such as the current
> "vm-enabled", "vm-mailto", "vm-mailfrom" parameters in voice-message.
>
> It will be much easy to implement fax2email function. And much more c
> oupling.
>
> Most of the email fuctions should be the same for both modules.
>
> Thanks!
>
> 2012-11-27
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*Abaci
> *?????*2012-11-27 02:57
> *???*Re: [Freeswitch-dev] About module "Mod_pandsp"
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*
>
>  I think what he's looking for is automatic fax to email like Voicemail
> to email is handled. this is not something FreeSWITCH has, not sure if it's
> because of related patents or something else.
>
> On 11/26/2012 9:58 AM, sparklezou wrote:
>
> **
> Hi Mod_spandsp developers,
>
> Is it possible to develop Mod_spandsp like "Mod_voicemail"?
>
> I means, set all of the related parameters in the user configure file.
> such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail
> sender configuration, such as SMTP.
>
> Something seemd the same, send "voice message" via eamil. Send "fax file"
> via email.
>
> Thanks!
>
> 2012-11-26
> ------------------------------
>  sparklezou
> **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From abaci64 at gmail.com  Tue Dec 18 19:35:30 2012
From: abaci64 at gmail.com (Abaci)
Date: Tue, 18 Dec 2012 11:35:30 -0500
Subject: [Freeswitch-dev] About module "Mod_pandsp",
	send the fax by email
In-Reply-To: <6bfa4541.304e.13bac095d0c.Coremail.sparklezou@163.com>
References: <50B3BB83.60700@gmail.com><50B3FEC6.2060709@163.com>
	<6bfa4541.304e.13bac095d0c.Coremail.sparklezou@163.com>
Message-ID: <50D09B52.8000809@gmail.com>

Not sure why you keep on asking the same question. you can either do 
what Michael Collins suggested (make a script), If you insist on having 
FreeSWITCh do it you can try to open a feature request on Jira with a 
Bounty, and see if someone is interested in doing it.

On 12/17/2012 10:20 PM, sparklezou wrote:
> Hi All,
> I would like to implement the following feature on fax module 
> "Mod_pandsp".
> Fax-to-Eaml, this feature should be simular to "Voicemail-to-Email".
> 1. hardware fax detection, then invoke "Mod_pandsp" at FreeSwitch 
> side. Then send the received fax to the individual email. The email 
> info should be set in the user account.
> 2.software fax detection, detected by "spandsp_start_fax_detect". But 
> it should "answer" the call firstly.
> 3. The caller part side wait for a fax tone manually, then start to 
> transfer the fax. Answer the call as common. Then press "*" or "#" to 
> send fax tone to receive fax by email.
> Email-to-Fax
> 1. send the document which should be faxed as attachment. and in the 
> email title, it's the destination number.
> 2. Freeswitch will check the mailbox timely. send the fax. Then reply 
> one email, success or NOT.
> Thanks!
> 2012-12-18
> ------------------------------------------------------------------------
> sparklezou
> ------------------------------------------------------------------------
> *???:*sparklezou
> *????:*2012-11-27 07:52
> *??:*Re: Re: [Freeswitch-dev] About module "Mod_pandsp"
> *???:*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *??:*
> Hi All,
> Correct!
> I hope the emaill function for voice-message and fax file coulde be 
> combind.
> Both "Mod_spandsp" and "Mod_voicemail" could get the usre email 
> configuration from the user profile. such as the current 
> "vm-enabled", "vm-mailto", "vm-mailfrom" parameters in voice-message.
> It will be much easy to implement fax2email function. And much more 
> coupling.
> Most of the email fuctions should be the same for both modules.
> Thanks!
> 2012-11-27
> ------------------------------------------------------------------------
> sparklezou
> ------------------------------------------------------------------------
> *???:*Abaci
> *????:*2012-11-27 02:57
> *??:*Re: [Freeswitch-dev] About module "Mod_pandsp"
> *???:*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *??:*
> I think what he's looking for is automatic fax to email like Voicemail 
> to email is handled. this is not something FreeSWITCH has, not sure if 
> it's because of related patents or something else.
>
> On 11/26/2012 9:58 AM, sparklezou wrote:
>> Hi Mod_spandsp developers,
>> Is it possible to develop Mod_spandsp like "Mod_voicemail"?
>> I means, set all of the related parameters in the user configure 
>> file. such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. 
>> Also the mail sender configuration, such as SMTP.
>> Something seemd the same, send "voice message" via eamil. Send "fax 
>> file" via email.
>> Thanks!
>> 2012-11-26
>> ------------------------------------------------------------------------
>> sparklezou
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From orn at arnarson.net  Tue Dec 18 18:18:38 2012
From: orn at arnarson.net (=?UTF-8?Q?=C3=96rn_Arnarson?=)
Date: Tue, 18 Dec 2012 15:18:38 +0000
Subject: [Freeswitch-dev] A bug in dealing with buggy DTMF handling?
Message-ID: <CAG_7Y1QwrubUSjgs6ENuXjJoSHbu7=sJoqTmUUedxXZmrtiBew@mail.gmail.com>

Hello,

I'm dealing with an end device whose designers don't seem to grasp the
concept of negotiation.

The device in question "supports" inband DTMF as well as RFC2833. However,
once you set it to either, it will blindly use it, regardless of the
support of the other end.

I have a scenario now where I set dtmf_type=none, and export dtmf_type=none
in the dialplan for certain calls, and this works beautifully for devices
that actually listen and don't transmit RFC2833 when it's not an option in
the SDP. However, when the device sends RFC2833 in spite of it not being
supported as per the SDP, FreeSwitch will send the DTMF to the b-leg as
SIP-INFO, despite dtmf_type having been set to none.

If I export dtmf_type=rfc2833, it will correctly use RFC2833 for the b-leg.
If I do not export it at all it will also use RFC2833.

Most likely, it's not intended for FS to use SIP-INFO when explicitly told
to use inband, even though the device on the A-leg is buggy.

Any thoughts on the matter? Is this a conscious choice?

I'm running FS stable, version 1.2.3.

Regards,
?rn
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From sparklezou at 163.com  Wed Dec 19 04:16:30 2012
From: sparklezou at 163.com (sparklezou)
Date: Wed, 19 Dec 2012 09:16:30 +0800
Subject: [Freeswitch-dev] About module "Mod_pandsp",
	send the fax by email
In-Reply-To: <50D09B52.8000809@gmail.com>
References: <50D09B52.8000809@gmail.com>
Message-ID: <12cf402b.f07.13bb0bd5ac8.Coremail.sparklezou@163.com>

Hi,

At start, I just reviewed the modules drafftly. Now I have finish such test usage of these modules. And consider the real situation.

I also have read some script in the wiki. I think it should be better to interaged into Freeswitch.

And the email function should be better independ as the basic function module. 

I'm still studying the Freeswitch. I would like to contribute my idea. And do some basic coding work at current stage.

If I could use Freeswitch into one project, and could get money, I'm glade to donate the benifit to Freeswitch also. :-)

Thanks!

2012-12-19



sparklezou



????Abaci
?????2012-12-19 00:35
???Re: [Freeswitch-dev] About module "Mod_pandsp", send the fax by email
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Not sure why you keep on asking the same question. you can either do what Michael Collins suggested (make a script), If you insist on having FreeSWITCh do it you can try to open a feature request on Jira with a Bounty, and see if someone is interested in doing it.

On 12/17/2012 10:20 PM, sparklezou wrote:

Hi All,

I would like to implement the following feature on fax module "Mod_pandsp".

Fax-to-Eaml, this feature should be simular to "Voicemail-to-Email".

1. hardware fax detection, then invoke "Mod_pandsp" at FreeSwitch side. Then send the received fax to the individual email. The email info should be set in the user account.

2.software fax detection, detected by "spandsp_start_fax_detect". But it should "answer" the call firstly.

3. The caller part side wait for a fax tone manually, then start to transfer the fax. Answer the call as common. Then press "*" or "#" to send fax tone to receive fax by email.


Email-to-Fax

1. send the document which should be faxed as attachment. and in the email title, it's the destination number. 

2. Freeswitch will check the mailbox timely. send the fax. Then reply one email, success or NOT.

Thanks!

2012-12-18



sparklezou



????sparklezou
?????2012-11-27 07:52
???Re: Re: [Freeswitch-dev] About module "Mod_pandsp"
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

Hi All,

Correct! 

I hope the emaill function for voice-message and fax file coulde be combind.

Both "Mod_spandsp" and "Mod_voicemail" could get the usre email configuration from the user profile. such as the current "vm-enabled", "vm-mailto", "vm-mailfrom" parameters in voice-message.

It will be much easy to implement fax2email function. And much more coupling.

Most of the email fuctions should be the same for both modules.

Thanks!

2012-11-27



sparklezou



????Abaci
?????2012-11-27 02:57
???Re: [Freeswitch-dev] About module "Mod_pandsp"
????"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
???

I think what he's looking for is automatic fax to email like Voicemail to email is handled. this is not something FreeSWITCH has, not sure if it's because of related patents or something else.

On 11/26/2012 9:58 AM, sparklezou wrote:

Hi Mod_spandsp developers,

Is it possible to develop Mod_spandsp like "Mod_voicemail"?

I means, set all of the related parameters in the user configure file. such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail sender configuration, such as SMTP.

Something seemd the same, send "voice message" via eamil. Send "fax file" via email.

Thanks!

2012-11-26



sparklezou

 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org




 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
http://www.freeswitch.org
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From msc at freeswitch.org  Wed Dec 19 05:23:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Dec 2012 18:23:36 -0800
Subject: [Freeswitch-dev] About module "Mod_pandsp",
	send the fax by email
In-Reply-To: <12cf402b.f07.13bb0bd5ac8.Coremail.sparklezou@163.com>
References: <50D09B52.8000809@gmail.com>
	<12cf402b.f07.13bb0bd5ac8.Coremail.sparklezou@163.com>
Message-ID: <CAKzWOxWUWou6zprQVVAT0mLiT=yBRdxCRACWy+e8u3WYtgGEJg@mail.gmail.com>

If you can make it work by editing the source code then go for it! The gang
here will try to help wherever they reasonably can.
-MC

On Tue, Dec 18, 2012 at 5:16 PM, sparklezou <sparklezou at 163.com> wrote:

> **
> Hi,
>
> At start, I just reviewed the modules drafftly. Now I have finish such
> test usage of these modules. And consider the real situation.
>
> I also have read some script in the wiki. I think it should be better to
> interaged into Freeswitch.
>
> And the email function should be better independ as the basic function
> module.
>
> I'm still studying the Freeswitch. I would like to contribute my idea.
> And do some basic coding work at current stage.
>
> If I could use Freeswitch into one project, and could get money, I'm glade
> to donate the benifit to Freeswitch also. :-)
>
> Thanks!
>
> 2012-12-19
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*Abaci
> *?????*2012-12-19 00:35
> *???*Re: [Freeswitch-dev] About module "Mod_pandsp", send the fax by email
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org>
> *???*
>
>  Not sure why you keep on asking the same question. you can either do
> what Michael Collins suggested (make a script), If you insist on having
> FreeSWITCh do it you can try to open a feature request on Jira with a
> Bounty, and see if someone is interested in doing it.
>
> On 12/17/2012 10:20 PM, sparklezou wrote:
>
> Hi All,
>
> I would like to implement the following feature on fax module "Mod_pandsp".
>
> Fax-to-Eaml, this feature should be simular to "Voicemail-to-Email".
>
> 1. hardware fax detection, then invoke "Mod_pandsp" at FreeSwitch side.
> Then send the received fax to the individual email. The email info should
> be set in the user account.
>
> 2.software fax detection, detected by "spandsp_start_fax_detect". But it
> should "answer" the call firstly.
>
> 3. The caller part side wait for a fax tone manually, then start to
> transfer the fax. Answer the call as common. Then press "*" or "#" to send
> fax tone to receive fax by email.
>
>
> Email-to-Fax
>
> 1. send the document which should be faxed as attachment. and in the email
> title, it's the destination number.
>
> 2. Freeswitch will check the mailbox timely. send the fax. Then reply one
> email, success or NOT.
>
> Thanks!
>
> 2012-12-18
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*sparklezou
> *?????*2012-11-27 07:52
> *???*Re: Re: [Freeswitch-dev] About module "Mod_pandsp"
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org><freeswitch-dev at lists.freeswitch.org>
> *???*
>
>  Hi All,
>
> Correct!
>
> I hope the emaill function for voice-message and fax file coulde be
> combind.
>
> Both "Mod_spandsp" and "Mod_voicemail" could get the usre email
> configuration from the user profile. such as the current
> "vm-enabled", "vm-mailto", "vm-mailfrom" parameters in voice-message.
>
> It will be much easy to implement fax2email function. And much more c
> oupling.
>
> Most of the email fuctions should be the same for both modules.
>
> Thanks!
>
> 2012-11-27
>  ------------------------------
>  sparklezou
>  ------------------------------
>  *????*Abaci
> *?????*2012-11-27 02:57
> *???*Re: [Freeswitch-dev] About module "Mod_pandsp"
> *????*"freeswitch-dev"<freeswitch-dev at lists.freeswitch.org><freeswitch-dev at lists.freeswitch.org>
> *???*
>
>  I think what he's looking for is automatic fax to email like Voicemail
> to email is handled. this is not something FreeSWITCH has, not sure if it's
> because of related patents or something else.
>
> On 11/26/2012 9:58 AM, sparklezou wrote:
>
> **
> Hi Mod_spandsp developers,
>
> Is it possible to develop Mod_spandsp like "Mod_voicemail"?
>
> I means, set all of the related parameters in the user configure file.
> such as "vm-enabled", "vm-mailto", "vm-mailfrom", and so on. Also the mail
> sender configuration, such as SMTP.
>
> Something seemd the same, send "voice message" via eamil. Send "fax file"
> via email.
>
> Thanks!
>
> 2012-11-26
> ------------------------------
>  sparklezou
> **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
>
> FreeSWITCH-dev mailing listFreeSWITCH-dev at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-devhttp://www.freeswitch.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 19 20:19:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Dec 2012 09:19:44 -0800
Subject: [Freeswitch-dev] FreeSWITCH Weekly Conference Call Today
Message-ID: <CAKzWOxVo=-xZ=oEdnByLuGpyKevL_hJqxqFS5v0a=t5WqYJz5A@mail.gmail.com>

Hello all!

We have the weekly conference call today:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_19

We'll be taking a look at the new nested conditions feature that was
recently added to the XML dialplan.

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Wed Dec 26 20:03:36 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Dec 2012 09:03:36 -0800
Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today
Message-ID: <CAKzWOxXFSYwXoCQSp6hY=RebbdqxoXH-6FEfqj5EMZO_nmJQGg@mail.gmail.com>

Hi gang!

I know a lot of folks are gone for the holidays but anyone who is around is
welcome to join us today. We have a light agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_12_26

However, I believe at least one person will be joining us to talk about a
potential job for a FreeSWITCH contractor. Hope to talk to you all soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From juanito1982 at gmail.com  Thu Dec 27 20:47:44 2012
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 27 Dec 2012 18:47:44 +0100
Subject: [Freeswitch-dev] Install precompiled module
Message-ID: <CAHLYDCtbYfqPijBJK826HgfvVKwuugM8buP3W52aPtELW_UA_w@mail.gmail.com>

Hello,

   I'd like to know which steps must be made to install/update one module
compiled inside a box different from the running one. Both boxes runs same
SO and FS versions.

Regards
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From msc at freeswitch.org  Thu Dec 27 21:02:58 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 10:02:58 -0800
Subject: [Freeswitch-dev] Install precompiled module
In-Reply-To: <CAHLYDCtbYfqPijBJK826HgfvVKwuugM8buP3W52aPtELW_UA_w@mail.gmail.com>
References: <CAHLYDCtbYfqPijBJK826HgfvVKwuugM8buP3W52aPtELW_UA_w@mail.gmail.com>
Message-ID: <CAKzWOxWx42-f=yTzXZMaCRknjDLTpXTQQuUusMfu1bjiCtGopw@mail.gmail.com>

If both boxes are the same platform then you can compile on box A and copy
over to box B. Usually you will have files in /usr/local/freeswitch/mod.
Copy over the .so and .la files from A to B for the module in question,
then try doing "load mod_xxx" at the fs_cli on B.

-MC

On Thu, Dec 27, 2012 at 9:47 AM, Juan Antonio Iba?ez Santorum <
juanito1982 at gmail.com> wrote:

> Hello,
>
>    I'd like to know which steps must be made to install/update one module
> compiled inside a box different from the running one. Both boxes runs same
> SO and FS versions.
>
> Regards
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From steveayre at gmail.com  Thu Dec 27 22:23:27 2012
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 27 Dec 2012 19:23:27 +0000
Subject: [Freeswitch-dev] Install precompiled module
In-Reply-To: <CAKzWOxWx42-f=yTzXZMaCRknjDLTpXTQQuUusMfu1bjiCtGopw@mail.gmail.com>
References: <CAHLYDCtbYfqPijBJK826HgfvVKwuugM8buP3W52aPtELW_UA_w@mail.gmail.com>
	<CAKzWOxWx42-f=yTzXZMaCRknjDLTpXTQQuUusMfu1bjiCtGopw@mail.gmail.com>
Message-ID: <91F72020-7C86-48E4-A11F-92333F97911F@gmail.com>

On Debian look at the libfreeswitch-dev module for building redistributable .deb packages - I'll put some information on this in the wiki in the near future.

On CentOS a similar RPM should be possible.

Steve



On 27 Dec 2012, at 18:02, Michael Collins <msc at freeswitch.org> wrote:

> If both boxes are the same platform then you can compile on box A and copy over to box B. Usually you will have files in /usr/local/freeswitch/mod. Copy over the .so and .la files from A to B for the module in question, then try doing "load mod_xxx" at the fs_cli on B.
> 
> -MC
> 
> On Thu, Dec 27, 2012 at 9:47 AM, Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com> wrote:
>> Hello,
>> 
>>    I'd like to know which steps must be made to install/update one module compiled inside a box different from the running one. Both boxes runs same SO and FS versions.
>> 
>> Regards
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-dev mailing list
>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>> http://www.freeswitch.org
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
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From msc at freeswitch.org  Fri Dec 28 01:40:59 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Dec 2012 14:40:59 -0800
Subject: [Freeswitch-dev] Need community input: hangup_hook recipes
Message-ID: <CAKzWOxWBcGOuQBxkCNMCxqpPPBEF__GwEs4qmjuuJN14BSU7dA@mail.gmail.com>

Hey all,

I was working on a feature for a friend of mine and it turns out that it
might be useful to put on the wiki. I noticed that we don't have a place
explicitly for hangup hook recipes. I was thinking about making a page just
for "interesting things you can do when the call ends" and then linking to
it from the dialplan recipes page.

Two questions: does that sound like a good place to put these recipes? does
anyone have any recipes they'd like to contribute?

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From dxj19831029 at gmail.com  Mon Dec 31 16:33:18 2012
From: dxj19831029 at gmail.com (Xijing Dai)
Date: Mon, 31 Dec 2012 21:33:18 +0800
Subject: [Freeswitch-dev] Freeswitch-----multiple channels support(fund)
Message-ID: <CAMsCbX9TjQMTfqgv6Gktn_3tRi7p1ptWb33ALg5anvFGooMrHQ@mail.gmail.com>

Hey all,


We are looking for someone who want to add multiple channels (stereo at
least) to freeswitch. Plus 44100 sample rate.

We can provide some fund.

If you think you can do it for the community, tell us how much you need. If
we can afford, we would :).


By the way, Happy new year!


Cheers
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